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58
BUGS
58
BUGS
@@ -1,22 +1,48 @@
|
||||
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:
|
||||
* 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.
|
||||
|
||||
http://bugs.digium.com
|
||||
* When you flip to call waiting on a tormenta channel while you have a
|
||||
three way call up, the parties in the three way cannot hear one another
|
||||
in the general case.
|
||||
|
||||
For more information on using the bug tracker, or to
|
||||
learn how you can contribute by acting as a bug marshall
|
||||
please see:
|
||||
* No auto-reload in chan_zap yet
|
||||
|
||||
http://www.asterisk.org/developers/bug-guidelines
|
||||
* Must be able to call park with flash-hook transfer
|
||||
|
||||
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:
|
||||
======================================================================
|
||||
Short report on the voicemail system
|
||||
======================================================================
|
||||
Stuff We Need:
|
||||
|
||||
http://lists.digium.com
|
||||
|
||||
Thank you!
|
||||
-Date/Time (conversion on the fly for different locales)
|
||||
-A more fleshed/emphasized Main Menu
|
||||
-Mailbox Options
|
||||
-useful for allowing user to set certain options
|
||||
-Notification of new vs. old messages
|
||||
-Notification of first and last messages (old and new)
|
||||
and a return to the Main Menu @ the end
|
||||
**-Better handling of lack of user input, specifically...
|
||||
infinite loops...
|
||||
currently found in: vm-instructions
|
||||
vm-msginstruct
|
||||
System MUST disconnect user for inactivity
|
||||
-Mid message menu w/
|
||||
pause/unpause
|
||||
seeking
|
||||
callback option
|
||||
option to get caller's number if available
|
||||
option to leave message directly on caller's voicemail
|
||||
if he/she has account on system
|
||||
-Also redesign the End Of Message Menu
|
||||
-Efficienty Question...
|
||||
Better to always rename msgs to being @ 0001
|
||||
or...
|
||||
Better to append new msgs numerically @ the
|
||||
end and use software to traverse them in
|
||||
order...saving cpu cycles on renaming files
|
||||
..could get messy w/ lots of users
|
||||
|
||||
486
CHANGES
486
CHANGES
@@ -1,341 +1,145 @@
|
||||
Changes since Asterisk 1.2:
|
||||
|
||||
* over 4,000 commits since 1.2
|
||||
* queue member naming
|
||||
* CLI commands rework
|
||||
o Change the way CLI commands are structured.
|
||||
o Most commands are now <module> <verb> <args>
|
||||
* chan_h323 update
|
||||
* RTP packetization
|
||||
* SLA (Shared Line Appearance) support
|
||||
* T.38 Passthrough Support for faxing in SIP
|
||||
* Generic channel jitterbuffer (spawned from RTP)
|
||||
* Variable Length DTMF for better DTMF compatibility
|
||||
* Improved chan_iax2 scalability by using multithreading
|
||||
* AEL2 has replaced the original implementation of AEL. The "2" is removed. For more details,
|
||||
read: http://www.voip-info.org/wiki/view/Asterisk+AEL2
|
||||
AEL is no longer considered experimental.
|
||||
* New sounds; English, Spanish, and French prompts, as well as music on hold files, in
|
||||
multiple Asterisk native formats.
|
||||
* IMAP storage of voicemail
|
||||
* Jabber/GoogleTalk integration
|
||||
* New speech recognition API for interfacing to different Voice Recognition software packages
|
||||
* much more customizable and portable build system
|
||||
o also for asterisk-addons
|
||||
* Radius CDR logging
|
||||
* SNMP support
|
||||
* SMDI (Simplified Message Desk Interface) support
|
||||
* Redesign of MusicOnHold configuration settings
|
||||
* Manager over HTTP
|
||||
* Significant chan_skinny updates
|
||||
* Significant chan_misdn updates
|
||||
* Improved SIP transfers
|
||||
* SIP MWI subscription support
|
||||
* Much improved support for SIP video
|
||||
* Control over SIP transfers and subscriptions (enable/disable per device)
|
||||
* ChanSpy whisper mode (Whisper Paging)
|
||||
* Configurable language support for saying dates and times
|
||||
* Significant architecture improvements for memory usage and performance
|
||||
* Media-only IAX2 transfers
|
||||
* Updates to the Radio Repeater app code
|
||||
* Deprecation of AgentCallbackLogin in favor of a dialplan-based solution
|
||||
* uClibc builds supported
|
||||
* Work done for freeBSD portability
|
||||
* Work done for Solaris portability
|
||||
* FreeTDS-based database can be used with Realtime
|
||||
* New internal data structure, stringfields, is implemented in IAX and SIP, reducing memory consumption by about 50%.
|
||||
* Use of thread local storage for reduced memory allocation/freeing and lower stack consumption
|
||||
* Reorganized files into docs/ main/ configs/, including name changes in some cases
|
||||
* Much effort was expended in arranging documentation in source files in doxygen format
|
||||
* Improved IP TOS support for IAX and SIP
|
||||
* Builtin mini HTTP server
|
||||
* Added support for Sigma Designs cards.
|
||||
* Frame header caching to reduce memory allocation/freeing
|
||||
* Passthrough and record/playback support for G.722 wideband audio
|
||||
* using mpg123 to play MP3 files for music-on-hold will be deprecated in 1.4 (start using the "native support")
|
||||
* New Apps:
|
||||
1. AMD() ;; Answering Machine Detection
|
||||
2. ChannelRedirect() ;; asynch goto, redirect chan to context/exten/priority
|
||||
3. ContinueWhile() ;; Addition to the While() suite. Acts like "continue".
|
||||
4. ExitWhile() ;; Addition to the While() suite. Acts like "break".
|
||||
5. ExtenSpy() ;; A close cousin to ChanSpy().
|
||||
6. FollowMe() ;; findme/followme call redirect app
|
||||
7. Log() ;; Send a message to the log, based on severity level.
|
||||
8. MacroExclusive() ;; No more than one invocation of this macro allowed at any one time.
|
||||
9. MorseCode() ;; turns strings into dits and dahs. A playground for ham radio licensees!
|
||||
10. OSPAuth() ;; OSP authentication
|
||||
11. QueueLog() ;; allows you to write your own events into the queue log
|
||||
12. SLAStation() ;; Shared Line Appearance
|
||||
13. SLATrunk() ;; Shared Line Appearance
|
||||
14. SpeechCreate() ;; Voice Recognition Engine interface...
|
||||
15. SpeechActivateGrammar()
|
||||
16. SpeechStart()
|
||||
17. SpeechBackground
|
||||
18. SpeechDeactivateGrammar()
|
||||
19. SpeechProcessingSound()
|
||||
20. SpeechDestroy()
|
||||
21. SpeechLoadGrammar()
|
||||
22. SpeechUnloadGrammar()
|
||||
23. StopMixMonitor() ;; to stop the MixMonitor App.
|
||||
24. TryExec() ;; execute dialplan app without fatal consequences
|
||||
* Apps removed:
|
||||
1. CheckGroup -- do a comparison to ${GROUP()}
|
||||
2. Curl -- use the function CURL() instead
|
||||
3. Cut -- use the function CUT() instead
|
||||
4. DateTime -- use sayunixtime() app instead.
|
||||
5. DBget -- deprecated in 1.2, now removed.
|
||||
6. DBput -- deprecated in 1.2, now removed.
|
||||
7. Enumlookup -- use the function ENUMLOOKUP() instead
|
||||
8. Eval -- use the function EVAL() instead
|
||||
9. GetGroupCount -- use the function GROUP_COUNT() instead
|
||||
10. GetGroupMatchCount -- use the function GROUP_MATCH_COUNT() instead
|
||||
11. Intercom -- use the chan_oss module instead
|
||||
12. Math -- use the function MATH() instead
|
||||
13. MD5 -- use the function MD5() instead
|
||||
14. SetCIDname -- use the function CALLERID(name) instead
|
||||
15. SetCIDnum -- use the function CALLERID(number) instead
|
||||
16. SetGroup -- use Set(GROUP=group) instead
|
||||
17. SetRDNIS -- use the function CALLERID(rdnis) instead
|
||||
18. Sql_postgres -- was deprecated in 1.2, now removed
|
||||
19. Txtcidname -- use the function TXTCIDNAME instead
|
||||
* New Dialplan Functions:
|
||||
1. ARRAY()
|
||||
2. BASE_64_DECODE()
|
||||
3. BASE_64_ENCODE()
|
||||
4. CHANNEL()
|
||||
5. CURL()
|
||||
6. CUT()
|
||||
7. DB_DELETE()
|
||||
8. FILTER()
|
||||
9. GLOBAL()
|
||||
10. IFTIME()
|
||||
11. KEYPADHASH()
|
||||
12. ODBC()
|
||||
13. QUOTE()
|
||||
14. RAND()
|
||||
15. REALTIME()
|
||||
16. SHA1()
|
||||
17. SORT()
|
||||
18. SPRINTF()
|
||||
19. SQL_ESC()
|
||||
20. STAT()
|
||||
21. STRPTIME()
|
||||
* Apps that have changes to their interface:
|
||||
1. Authenticate() -- optional maxdigits argument added.
|
||||
2. ChanSpy() -- new options:
|
||||
o w -- Enable 'whisper' mode, so the spying channel can talk to...
|
||||
o W -- Enable 'private whisper' mode, so the spying channel can...
|
||||
3. DBdel() -- now marked as DEPRECATED in favor of the DB_DELETE func
|
||||
4. Dial()
|
||||
o New Option: O([x]) for Zaptel operator mode
|
||||
o New Option: K/k parking via dtmf tones
|
||||
5. Dictate() -- optional filename argument added.
|
||||
6. Directory() -- new option: e - In addition to the name, also read the extension number...
|
||||
7. Meetme() -- new options:
|
||||
o 'I' -- announce user join/leave without review
|
||||
o 'l' -- set listen only mode (Listen only, no talking)
|
||||
o 'o' -- set talker optimization - treats talkers who aren't speaking as...
|
||||
o '1' -- do not play message when first person enters
|
||||
8. MeetmeAdmin() -- new options:
|
||||
o 'r' -- Reset one user's volume settings
|
||||
o 'R' -- Reset all users volume settings
|
||||
o 's' -- Lower entire conference speaking volume
|
||||
o 'S' -- Raise entire conference speaking volume
|
||||
o 't' -- Lower one user's talk volume
|
||||
o 'T' -- Lower all users talk volume
|
||||
o 'u' -- Lower one user's listen volume
|
||||
o 'U' -- Lower all users listen volume
|
||||
o 'v' -- Lower entire conference listening volume
|
||||
o 'V' -- Raise entire conference listening volume
|
||||
9. OSPFinish() : now also can return ERROR result.
|
||||
10. OSPLookup() : Sets more variables, also now returns ERROR result.
|
||||
11. Page() -- New option: r - record the page into a file (see 'r' for app_meetme)
|
||||
12. Pickup() -- multiple extensions, PICKUPMARK; read the description!
|
||||
13. Queue()
|
||||
o New Argument: AGI
|
||||
o New option: i
|
||||
14. Random() -- is now deprecated in 1.4
|
||||
15. Read() -- replace 'skip' and 'noanswer' options with 's', 'n', add 'i' option.
|
||||
16. Record() -- New option: 'x' : ignore all terminator keys (DTMF) and keep recording until hangup
|
||||
17. UserEvent() -- slight change in behavior. Read the description.
|
||||
18. VoiceMailMain() -- new a(#) option, goes to folder # directly.
|
||||
19. WaitForSilence() -- new optional 3rd arg, time delay before returning.
|
||||
* Functions that have changes to their interfaces:
|
||||
1. CDR -- new option: u
|
||||
2. LANGUAGE -- Deprecated. Use CHANNEL(language) instead.
|
||||
3. MUSICCLASS -- Deprecated. Use CHANNEL(musicclass) instead.
|
||||
* Configuration File Changes:
|
||||
1. NEW config files:
|
||||
1. amd.conf -- Answering Machine Detection parameters
|
||||
2. followme.conf -- parameters for the findme/followme call forwarding
|
||||
3. func_odbc.conf -- define sql access functions here
|
||||
4. gtalk.conf -- how to handle gtalk protocol calls
|
||||
5. h323.conf -- h323 configuration
|
||||
6. http.conf -- config for the builtin mini-http server in asterisk
|
||||
7. jabber.conf -- jabber interface
|
||||
8. jingle.conf -- jingle protocol interface config
|
||||
10. res_snmp.conf -- to enable snmp in asterisk, and define full/sub agent status
|
||||
11. say.conf -- define per-language rules for numbers, dates, etc.
|
||||
12. skinny.conf -- for those special skinny phones you want to use...
|
||||
13. sla.conf -- Shared Line Appearance config
|
||||
14. smdi.conf -- SMDI messaging config
|
||||
15. udptl.conf -- T38's udptl transport config
|
||||
16. users.conf -- user config
|
||||
2. Changes to Existing Config files:
|
||||
1. In General:
|
||||
o Jitterbuffer support added to several channels. Usually adds these variables to a config file:
|
||||
1. jbenable
|
||||
2. jbmaxsize
|
||||
3. jbresyncthreshold
|
||||
4. jbimpl
|
||||
5. jblog
|
||||
o MusicOnHold upgrade introduces two new variables:
|
||||
1. mohinterpret
|
||||
2. mohsuggest
|
||||
2. agents.conf
|
||||
o maxlogintries variable added
|
||||
o autologoffunavail variable added
|
||||
o endcall variable added
|
||||
o agentgoodbye variable added
|
||||
o createlink variable REMOVED
|
||||
3. alsa.conf
|
||||
o mohinterpret variable added
|
||||
o Jitterbuffer variables added
|
||||
4. cdr.conf
|
||||
o endbeforehexten variable added
|
||||
o sections for csv and radius added, with variables usegmtime, loguniqueid,
|
||||
loguserfield, and radiuscfg variables.
|
||||
5. cdr_tds.conf
|
||||
o table variable added
|
||||
6. extensions.ael
|
||||
o Many upgrades. See the info at http://www.voip-info.org/wiki/view/Asterisk+AEL2
|
||||
7. extensions.conf
|
||||
o autofallthru now set to "yes" by default
|
||||
o userscontext variable added
|
||||
o added info/examples on paging and hints.
|
||||
8. features.conf
|
||||
o parkedplay variable added (who to beep at)
|
||||
o parkedmusicclass
|
||||
o atxfernoanswertimeout variable added
|
||||
o parkcall variable added (one step parking)
|
||||
o improved documentation for dynamic feature declarations!
|
||||
9. iax.conf
|
||||
o adsi variable added
|
||||
o mohinterpret variable added
|
||||
o mohsuggest variable added
|
||||
o jitterbuffer updates
|
||||
o iaxthreadcount variable added
|
||||
o iaxmaxthreadcount variable added
|
||||
o the way to specify TOS has changed.
|
||||
o mailboxdetail variable has been REMOVED.
|
||||
10. indications.conf
|
||||
o [bg] entry added (Bulgaria).
|
||||
o [il] entry added (Israel)
|
||||
o [in] entry added (India)
|
||||
o [jp] entry added (Japan)
|
||||
o [my] entry added (Malaysia)
|
||||
o [th] entry added (Thailand)
|
||||
11. manager.conf
|
||||
o webenabled variable added
|
||||
o httptimeout variable added
|
||||
o timestampevents variable added
|
||||
12. mgcp.conf
|
||||
o Jitterbuffer support added
|
||||
13. misdn.conf
|
||||
o l1watcher_timeout variable added
|
||||
o pp_l2_check variable added
|
||||
o echocancelwhenbridged variable added
|
||||
o echotraining variable added
|
||||
o max_incoming variable added
|
||||
o max_outgoing variable added
|
||||
14. modules.conf
|
||||
o a comment for preloading res_speech.so is added
|
||||
o mention of global symbols is removed
|
||||
o obsolesced entries for chan_modem_* and app_intercom have been removed
|
||||
15. musiconhold.conf
|
||||
o the default is now to do native moh from /var/lib/asterisk/moh
|
||||
16. osp.conf
|
||||
o authpolicy variable added
|
||||
17. oss.conf
|
||||
o debug variable added
|
||||
o device variable added
|
||||
o mixer variable added
|
||||
o boost variable added
|
||||
o callerid variable added
|
||||
o autohangup variable added
|
||||
o queuesize variable added
|
||||
o frags variable added
|
||||
o JitterBuffer support
|
||||
o sections to define alternate sound cards
|
||||
18. queues.conf
|
||||
o autofill variable added
|
||||
o monitor-type variable added
|
||||
o musiconhold is now musicclass, with a difference in interpretation
|
||||
o autofill variable added
|
||||
o autopause variable added
|
||||
o setinterfacevar variable added
|
||||
o ringinuse variable added
|
||||
19. res_odbc.conf
|
||||
o pooling variable added
|
||||
20. rpt.conf
|
||||
o duplex variable added
|
||||
o tailmessagetime variable added
|
||||
o tailsquashedtime variable added
|
||||
o tailmessages variable added
|
||||
21. rtp.conf
|
||||
o rtcpinterval varaible added
|
||||
22. sip.conf
|
||||
o allowoverlap variable added
|
||||
o allowtransfer variable added
|
||||
o tos variable REMOVED
|
||||
o tos_sip variable added
|
||||
o tos_audio variable added
|
||||
o tos_video variable added
|
||||
o minexpiry variable added
|
||||
o t1min variable added
|
||||
o musicclass variable REMOVED
|
||||
o mohinterpret variable added
|
||||
o maxcallbitratesuggest variable added
|
||||
o allowsubscribe variable added
|
||||
o videosupport variable added
|
||||
o maxcallbitrate variable added
|
||||
o g726nonstandard variable added
|
||||
o dumphistory variable added
|
||||
o allowsubscribe variable added
|
||||
o t38pt_udptl variable added
|
||||
o canreinvite variable can also now be set to 'nonat'
|
||||
o rtsavesysname variable added
|
||||
o JitterBuffer support added
|
||||
23. skinny.conf
|
||||
o port variable renamed to bindport
|
||||
o JitterBuffer support added
|
||||
o model variable REMOVED
|
||||
o mohinterpret variable added
|
||||
o mohsuggest variable added
|
||||
o speeddial variable added
|
||||
o addon variable added
|
||||
24. voicemail.conf
|
||||
o userscontext variable added
|
||||
o smdiport variable added
|
||||
o attachfmt variable added
|
||||
o volgain variable added
|
||||
o tempgreetwarn variable added
|
||||
25. zapata.conf
|
||||
o pritimer variable has improved documentation
|
||||
o New signalling method: fgccama
|
||||
o New signalling method: fgccamamf
|
||||
o outsignalling variable added
|
||||
o distinctiveringaftercid variable added
|
||||
o cidsignalling now also accepts v23_jp, and smdi
|
||||
o usesmdi variable added
|
||||
o smdiport variable added
|
||||
o mohinterpret variable added
|
||||
o mohsuggest variable added
|
||||
o JitterBuffer support added
|
||||
* Removed Codecs/Channels:
|
||||
1. codec_g723 was removed because the actual codec implementation it was designed to use is not distributable
|
||||
2. chan_modem_* and related modules are gone because the kernel support for those interfaces is old, buggy and unsupported
|
||||
* New Utils:
|
||||
1. aelparse -- compile .ael files outside of asterisk
|
||||
* New manager events:
|
||||
1. OriginateResponse event comes to replace OriginateSuccess and OriginateFailure
|
||||
Asterisk 0.1.11
|
||||
-- Add ISDN RAS capability
|
||||
-- Add stutter dialtone to Chan Zap
|
||||
-- Add "#include" capability to config files.
|
||||
-- Add call-forward variable to Chan Zap (*72, *73)
|
||||
-- Optimize IAX flow when transfer isn't possible
|
||||
-- Allow transmission of ANI over IAX
|
||||
Asterisk 0.1.10
|
||||
-- Make ast_readstring parameter be the max # of digits, not the max size with \0
|
||||
-- Make up any missing messages on the fly
|
||||
-- Add support for specific DTMF interruption to saying numbers
|
||||
-- Add new "u" and "b" options to condense busy/unavail handling
|
||||
-- Add support for RSA authentication on IAX calls
|
||||
-- Add support for ADSI compatible CPE
|
||||
-- Outgoing call queue
|
||||
-- Remote dialplan fixes for Quicknet
|
||||
-- Added AGI commands supporting TDD functions (RECEIVE CHAR & TDD MODE)
|
||||
-- Added TDD support (send/receive text in chan_zap)
|
||||
-- Fix all strncpy references
|
||||
-- Implement CSV CDR backend
|
||||
-- Implement Call Detail Records
|
||||
Asterisk 0.1.9
|
||||
-- Implement IAX quelching
|
||||
-- Allow Caller*ID to be overridden and suggested
|
||||
-- Configure defaults to use IAXTEL
|
||||
-- Allow remote dialplan polling via IAX
|
||||
-- Eliminate ast_longest_extension
|
||||
-- Implement dialplan request/reply
|
||||
-- Let peers have allow/disallow for codecs
|
||||
-- Change allow/deny to permit/deny in IAX
|
||||
-- Allow dialplan entries to match Caller*ID as well
|
||||
-- Added AGI (Asterisk Gateway Interface) scripting interface (app_agi)
|
||||
-- Added chan_zap for zapata telephony kernel interface, removed chan_tor
|
||||
-- Add convenience functions
|
||||
-- Fix race condition in channel hangup
|
||||
-- Fix memory leaks in both asterisk and iax frame allocations
|
||||
-- Add "iax show stats" command and -DTRACE_FRAMES (for frame tracing)
|
||||
-- Add DISA application (Thanks to Jim Dixon)
|
||||
-- Add IAX transfer support
|
||||
-- Add URL and HTML transmission
|
||||
-- Add application for sending images
|
||||
-- Add RedHat RPM spec file and build capability
|
||||
-- Fix GSM WAV file format bug
|
||||
-- Move ignorepat to main dialplan
|
||||
-- Add ability to specificy TOS bits in IAX
|
||||
-- Allow username:password in IAX strings
|
||||
-- Updates to PhoneJack interface
|
||||
-- Allow "servermail" in voicemail.conf to override e-mail in "from" line
|
||||
-- Add 'skip' option to app_playback
|
||||
-- Reject IAX calls on unknown extensions
|
||||
-- Fix version stuff
|
||||
Asterisk 0.1.8
|
||||
-- Keep track of version information
|
||||
-- Add -f to cause Asterisk not to fork
|
||||
-- Keep important information in voicemail .txt file
|
||||
-- Adtran Voice over Frame Relay updates
|
||||
-- Implement option setting/querying of channel drivers
|
||||
-- IAX performance improvements and protocol fixes
|
||||
-- Substantial enhancement of console channel driver
|
||||
-- Add IAX registration. Now IAX can dynamically register
|
||||
-- Add flash-hook transfer on tormenta channels
|
||||
-- Added Three Way Calling on tormenta channels
|
||||
-- Start on concept of zombie channel
|
||||
-- Add Call Waiting CallerID
|
||||
-- Keep track of who registeres contexts, includes, and extensions
|
||||
-- Added Call Waiting(tm), *67, *70, and *82 codes
|
||||
-- Move parked calls into "parkedcalls" context by default
|
||||
-- Allow dialplan to be displayed
|
||||
-- Allow "=>" instead of just "=" to make instantiation clearer
|
||||
-- Asterisk forks if called with no arguments
|
||||
-- Add remote control by running asterisk -vvvc
|
||||
-- Adjust verboseness with "set verbose" now
|
||||
-- No longer requires libaudiofile
|
||||
-- Install beep
|
||||
-- Make PBX Config module reload extensions on SIGHUP
|
||||
-- Allow modules to be reloaded when SIGHUP is received
|
||||
-- Variables now contain line numbers
|
||||
-- Make dialer send in band signalling
|
||||
-- Add record application
|
||||
-- Added PRI signalling to Tormenta driver
|
||||
-- Allow use of BYEXTENSION in "Goto"
|
||||
-- Allow adjustment of gains on tormenta channels
|
||||
-- Added raw PCM file format support
|
||||
-- Add U-law translator
|
||||
-- Fix DTMF handling in bridge code
|
||||
-- Fix access control with IAX
|
||||
* Asterisk 0.1.7
|
||||
-- Update configuration files and add some missing sounds
|
||||
-- Added ability to include one context in another
|
||||
-- Rewrite of PBX switching
|
||||
-- Major mods to dialler application
|
||||
-- Added Caller*ID spill reception
|
||||
-- Added Dialogic VOX file format support
|
||||
-- Added ADPCM Codec
|
||||
-- Add Tormenta driver (RBS signalling)
|
||||
-- Add Caller*ID spill creation
|
||||
-- Rewrite of translation layer entirely
|
||||
-- Add ability to run PBX without additional thread
|
||||
* Asterisk 0.1.6
|
||||
-- Make app_dial handle a lack of translators smoothly
|
||||
-- Add ISDN4Linux support -- dtmf is weird...
|
||||
-- Minor bug fixes
|
||||
* Asterisk 0.1.5
|
||||
-- Fix a small mistake in IAX
|
||||
-- Fix the QuickNet driver to work with newer cards
|
||||
* Asterisk 0.1.4
|
||||
-- Update VoFR some more
|
||||
-- Fix the QuickNet driver to work with LineJack
|
||||
-- Add ability to pass images for IAX.
|
||||
* Asterisk 0.1.3
|
||||
-- Update VoFR for latest sangoma code
|
||||
-- Update QuickNet Driver
|
||||
-- Add text message handling
|
||||
-- Fix transfers to use "default" if not in current context
|
||||
-- Add call parking
|
||||
-- Improve format/content negotiation
|
||||
-- Added support for multiple languages
|
||||
-- Bug fixes, as always...
|
||||
* Asterisk 0.1.2
|
||||
-- Updated README file with a "Getting Started" section
|
||||
-- Added sample sounds and configuration files.
|
||||
-- Added LPC10 very low bandwidth (low quality) compression
|
||||
-- Enhanced translation selection mechanism.
|
||||
-- Enhanced IAX jitter buffer, improved reliability
|
||||
-- Support echo cancelation on PhoneJack
|
||||
-- Updated PhoneJack driver to std. Telephony interface
|
||||
-- Added app_echo for evaluating VoIP latency
|
||||
-- Added app_system to execute arbitrary programs
|
||||
-- Updated sample configuration files
|
||||
-- Added OSS channel driver (full duplex only)
|
||||
-- Added IAX implementation
|
||||
-- Fixed some deadlocks.
|
||||
-- A whole bunch of bug fixes
|
||||
* Asterisk 0.1.1
|
||||
-- Revised translator, fixed some general race conditions throughout *
|
||||
-- Made dialer somewhat more aware of incompatible voice channels
|
||||
-- Added Voice Modem driver and A/Open Modem Driver stub
|
||||
-- Added MP3 decoder channel
|
||||
-- Added Microsoft WAV49 support
|
||||
-- Revised License -- Pure GPL, nothing else
|
||||
-- Modified Copyright statement since code is still currently owned by author
|
||||
-- Added RAW GSM headerless data format
|
||||
-- Innumerable bug fixes
|
||||
* Asterisk 0.1.0
|
||||
-- Initial Release
|
||||
|
||||
341
COPYING
341
COPYING
@@ -1,341 +0,0 @@
|
||||
|
||||
GNU GENERAL PUBLIC LICENSE
|
||||
Version 2, June 1991
|
||||
|
||||
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.
|
||||
|
||||
Preamble
|
||||
|
||||
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.
|
||||
|
||||
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.
|
||||
|
||||
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.
|
||||
|
||||
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.
|
||||
|
||||
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.
|
||||
|
||||
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.
|
||||
|
||||
Finally, any free program is threatened constantly by software
|
||||
patents. We wish to avoid the danger that redistributors of a free
|
||||
program will individually obtain patent licenses, in effect making the
|
||||
program proprietary. To prevent this, we have made it clear that any
|
||||
patent must be licensed for everyone's free use or not licensed at all.
|
||||
|
||||
The precise terms and conditions for copying, distribution and
|
||||
modification follow.
|
||||
|
||||
GNU GENERAL PUBLIC LICENSE
|
||||
TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
|
||||
|
||||
0. This License applies to any program or other work which contains
|
||||
a notice placed by the copyright holder saying it may be distributed
|
||||
under the terms of this General Public License. The "Program", below,
|
||||
refers to any such program or work, and a "work based on the Program"
|
||||
means either the Program or any derivative work under copyright law:
|
||||
that is to say, a work containing the Program or a portion of it,
|
||||
either verbatim or with modifications and/or translated into another
|
||||
language. (Hereinafter, translation is included without limitation in
|
||||
the term "modification".) Each licensee is addressed as "you".
|
||||
|
||||
Activities other than copying, distribution and modification are not
|
||||
covered by this License; they are outside its scope. The act of
|
||||
running the Program is not restricted, and the output from the Program
|
||||
is covered only if its contents constitute a work based on the
|
||||
Program (independent of having been made by running the Program).
|
||||
Whether that is true depends on what the Program does.
|
||||
|
||||
1. You may copy and distribute verbatim copies of the Program's
|
||||
source code as you receive it, in any medium, provided that you
|
||||
conspicuously and appropriately publish on each copy an appropriate
|
||||
copyright notice and disclaimer of warranty; keep intact all the
|
||||
notices that refer to this License and to the absence of any warranty;
|
||||
and give any other recipients of the Program a copy of this License
|
||||
along with the Program.
|
||||
|
||||
You may charge a fee for the physical act of transferring a copy, and
|
||||
you may at your option offer warranty protection in exchange for a fee.
|
||||
|
||||
2. You may modify your copy or copies of the Program or any portion
|
||||
of it, thus forming a work based on the Program, and copy and
|
||||
distribute such modifications or work under the terms of Section 1
|
||||
above, provided that you also meet all of these conditions:
|
||||
|
||||
a) You must cause the modified files to carry prominent notices
|
||||
stating that you changed the files and the date of any change.
|
||||
|
||||
b) You must cause any work that you distribute or publish, that in
|
||||
whole or in part contains or is derived from the Program or any
|
||||
part thereof, to be licensed as a whole at no charge to all third
|
||||
parties under the terms of this License.
|
||||
|
||||
c) If the modified program normally reads commands interactively
|
||||
when run, you must cause it, when started running for such
|
||||
interactive use in the most ordinary way, to print or display an
|
||||
announcement including an appropriate copyright notice and a
|
||||
notice that there is no warranty (or else, saying that you provide
|
||||
a warranty) and that users may redistribute the program under
|
||||
these conditions, and telling the user how to view a copy of this
|
||||
License. (Exception: if the Program itself is interactive but
|
||||
does not normally print such an announcement, your work based on
|
||||
the Program is not required to print an announcement.)
|
||||
|
||||
These requirements apply to the modified work as a whole. If
|
||||
identifiable sections of that work are not derived from the Program,
|
||||
and can be reasonably considered independent and separate works in
|
||||
themselves, then this License, and its terms, do not apply to those
|
||||
sections when you distribute them as separate works. But when you
|
||||
distribute the same sections as part of a whole which is a work based
|
||||
on the Program, the distribution of the whole must be on the terms of
|
||||
this License, whose permissions for other licensees extend to the
|
||||
entire whole, and thus to each and every part regardless of who wrote it.
|
||||
|
||||
Thus, it is not the intent of this section to claim rights or contest
|
||||
your rights to work written entirely by you; rather, the intent is to
|
||||
exercise the right to control the distribution of derivative or
|
||||
collective works based on the Program.
|
||||
|
||||
In addition, mere aggregation of another work not based on the Program
|
||||
with the Program (or with a work based on the Program) on a volume of
|
||||
a storage or distribution medium does not bring the other work under
|
||||
the scope of this License.
|
||||
|
||||
3. You may copy and distribute the Program (or a work based on it,
|
||||
under Section 2) in object code or executable form under the terms of
|
||||
Sections 1 and 2 above provided that you also do one of the following:
|
||||
|
||||
a) Accompany it with the complete corresponding machine-readable
|
||||
source code, which must be distributed under the terms of Sections
|
||||
1 and 2 above on a medium customarily used for software interchange; or,
|
||||
|
||||
b) Accompany it with a written offer, valid for at least three
|
||||
years, to give any third party, for a charge no more than your
|
||||
cost of physically performing source distribution, a complete
|
||||
machine-readable copy of the corresponding source code, to be
|
||||
distributed under the terms of Sections 1 and 2 above on a medium
|
||||
customarily used for software interchange; or,
|
||||
|
||||
c) Accompany it with the information you received as to the offer
|
||||
to distribute corresponding source code. (This alternative is
|
||||
allowed only for noncommercial distribution and only if you
|
||||
received the program in object code or executable form with such
|
||||
an offer, in accord with Subsection b above.)
|
||||
|
||||
The source code for a work means the preferred form of the work for
|
||||
making modifications to it. For an executable work, complete source
|
||||
code means all the source code for all modules it contains, plus any
|
||||
associated interface definition files, plus the scripts used to
|
||||
control compilation and installation of the executable. However, as a
|
||||
special exception, the source code distributed need not include
|
||||
anything that is normally distributed (in either source or binary
|
||||
form) with the major components (compiler, kernel, and so on) of the
|
||||
operating system on which the executable runs, unless that component
|
||||
itself accompanies the executable.
|
||||
|
||||
If distribution of executable or object code is made by offering
|
||||
access to copy from a designated place, then offering equivalent
|
||||
access to copy the source code from the same place counts as
|
||||
distribution of the source code, even though third parties are not
|
||||
compelled to copy the source along with the object code.
|
||||
|
||||
4. You may not copy, modify, sublicense, or distribute the Program
|
||||
except as expressly provided under this License. Any attempt
|
||||
otherwise to copy, modify, sublicense or distribute the Program is
|
||||
void, and will automatically terminate your rights under this License.
|
||||
However, parties who have received copies, or rights, from you under
|
||||
this License will not have their licenses terminated so long as such
|
||||
parties remain in full compliance.
|
||||
|
||||
5. You are not required to accept this License, since you have not
|
||||
signed it. However, nothing else grants you permission to modify or
|
||||
distribute the Program or its derivative works. These actions are
|
||||
prohibited by law if you do not accept this License. Therefore, by
|
||||
modifying or distributing the Program (or any work based on the
|
||||
Program), you indicate your acceptance of this License to do so, and
|
||||
all its terms and conditions for copying, distributing or modifying
|
||||
the Program or works based on it.
|
||||
|
||||
6. Each time you redistribute the Program (or any work based on the
|
||||
Program), the recipient automatically receives a license from the
|
||||
original licensor to copy, distribute or modify the Program subject to
|
||||
these terms and conditions. You may not impose any further
|
||||
restrictions on the recipients' exercise of the rights granted herein.
|
||||
You are not responsible for enforcing compliance by third parties to
|
||||
this License.
|
||||
|
||||
7. If, as a consequence of a court judgment or allegation of patent
|
||||
infringement or for any other reason (not limited to patent issues),
|
||||
conditions are imposed on you (whether by court order, agreement or
|
||||
otherwise) that contradict the conditions of this License, they do not
|
||||
excuse you from the conditions of this License. If you cannot
|
||||
distribute so as to satisfy simultaneously your obligations under this
|
||||
License and any other pertinent obligations, then as a consequence you
|
||||
may not distribute the Program at all. For example, if a patent
|
||||
license would not permit royalty-free redistribution of the Program by
|
||||
all those who receive copies directly or indirectly through you, then
|
||||
the only way you could satisfy both it and this License would be to
|
||||
refrain entirely from distribution of the Program.
|
||||
|
||||
If any portion of this section is held invalid or unenforceable under
|
||||
any particular circumstance, the balance of the section is intended to
|
||||
apply and the section as a whole is intended to apply in other
|
||||
circumstances.
|
||||
|
||||
It is not the purpose of this section to induce you to infringe any
|
||||
patents or other property right claims or to contest validity of any
|
||||
such claims; this section has the sole purpose of protecting the
|
||||
integrity of the free software distribution system, which is
|
||||
implemented by public license practices. Many people have made
|
||||
generous contributions to the wide range of software distributed
|
||||
through that system in reliance on consistent application of that
|
||||
system; it is up to the author/donor to decide if he or she is willing
|
||||
to distribute software through any other system and a licensee cannot
|
||||
impose that choice.
|
||||
|
||||
This section is intended to make thoroughly clear what is believed to
|
||||
be a consequence of the rest of this License.
|
||||
|
||||
8. If the distribution and/or use of the Program is restricted in
|
||||
certain countries either by patents or by copyrighted interfaces, the
|
||||
original copyright holder who places the Program under this License
|
||||
may add an explicit geographical distribution limitation excluding
|
||||
those countries, so that distribution is permitted only in or among
|
||||
countries not thus excluded. In such case, this License incorporates
|
||||
the limitation as if written in the body of this License.
|
||||
|
||||
9. The Free Software Foundation may publish revised and/or new versions
|
||||
of the General Public License from time to time. Such new versions will
|
||||
be similar in spirit to the present version, but may differ in detail to
|
||||
address new problems or concerns.
|
||||
|
||||
Each version is given a distinguishing version number. If the Program
|
||||
specifies a version number of this License which applies to it and "any
|
||||
later version", you have the option of following the terms and conditions
|
||||
either of that version or of any later version published by the Free
|
||||
Software Foundation. If the Program does not specify a version number of
|
||||
this License, you may choose any version ever published by the Free Software
|
||||
Foundation.
|
||||
|
||||
10. If you wish to incorporate parts of the Program into other free
|
||||
programs whose distribution conditions are different, write to the author
|
||||
to ask for permission. For software which is copyrighted by the Free
|
||||
Software Foundation, write to the Free Software Foundation; we sometimes
|
||||
make exceptions for this. Our decision will be guided by the two goals
|
||||
of preserving the free status of all derivatives of our free software and
|
||||
of promoting the sharing and reuse of software generally.
|
||||
|
||||
NO WARRANTY
|
||||
|
||||
11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY
|
||||
FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHEN
|
||||
OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES
|
||||
PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED
|
||||
OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
|
||||
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. THE ENTIRE RISK AS
|
||||
TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU. SHOULD THE
|
||||
PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING,
|
||||
REPAIR OR CORRECTION.
|
||||
|
||||
12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
|
||||
WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR
|
||||
REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES,
|
||||
INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING
|
||||
OUT OF THE USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED
|
||||
TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY
|
||||
YOU OR THIRD PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER
|
||||
PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE
|
||||
POSSIBILITY OF SUCH DAMAGES.
|
||||
|
||||
END OF TERMS AND CONDITIONS
|
||||
|
||||
How to Apply These Terms to Your New Programs
|
||||
|
||||
If you develop a new program, and you want it to be of the greatest
|
||||
possible use to the public, the best way to achieve this is to make it
|
||||
free software which everyone can redistribute and change under these terms.
|
||||
|
||||
To do so, attach the following notices to the program. It is safest
|
||||
to attach them to the start of each source file to most effectively
|
||||
convey the exclusion of warranty; and each file should have at least
|
||||
the "copyright" line and a pointer to where the full notice is found.
|
||||
|
||||
<one line to give the program's name and a brief idea of what it does.>
|
||||
Copyright (C) 19yy <name of author>
|
||||
|
||||
This program is free software; you can redistribute it and/or modify
|
||||
it under the terms of the GNU General Public License as published by
|
||||
the Free Software Foundation; either version 2 of the License, or
|
||||
(at your option) any later version.
|
||||
|
||||
This program is distributed in the hope that it will be useful,
|
||||
but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
||||
GNU General Public License for more details.
|
||||
|
||||
You should have received a copy of the GNU General Public License
|
||||
along with this program; if not, write to the Free Software
|
||||
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
|
||||
|
||||
Also add information on how to contact you by electronic and paper mail.
|
||||
|
||||
If the program is interactive, make it output a short notice like this
|
||||
when it starts in an interactive mode:
|
||||
|
||||
Gnomovision version 69, Copyright (C) 19yy name of author
|
||||
Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
|
||||
This is free software, and you are welcome to redistribute it
|
||||
under certain conditions; type `show c' for details.
|
||||
|
||||
The hypothetical commands `show w' and `show c' should show the appropriate
|
||||
parts of the General Public License. Of course, the commands you use may
|
||||
be called something other than `show w' and `show c'; they could even be
|
||||
mouse-clicks or menu items--whatever suits your program.
|
||||
|
||||
You should also get your employer (if you work as a programmer) or your
|
||||
school, if any, to sign a "copyright disclaimer" for the program, if
|
||||
necessary. Here is a sample; alter the names:
|
||||
|
||||
Yoyodyne, Inc., hereby disclaims all copyright interest in the program
|
||||
`Gnomovision' (which makes passes at compilers) written by James Hacker.
|
||||
|
||||
<signature of Ty Coon>, 1 April 1989
|
||||
Ty Coon, President of Vice
|
||||
|
||||
This General Public License does not permit incorporating your program into
|
||||
proprietary programs. If your program is a subroutine library, you may
|
||||
consider it more useful to permit linking proprietary applications with the
|
||||
library. If this is what you want to do, use the GNU Library General
|
||||
Public License instead of this License.
|
||||
171
CREDITS
171
CREDITS
@@ -1,177 +1,18 @@
|
||||
|
||||
=== 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 ===
|
||||
* Thanks to QuickNet Technologies for their donation of an Internet
|
||||
PhoneJack and Linejack card to the project. (http://www.quicknet.net)
|
||||
|
||||
* Thanks to VoipSupply for their donation of Sipura ATAs to the project for
|
||||
T.38 testing. (http://www.voipsupply.com)
|
||||
|
||||
* Thanks to Grandstream for their donation of ATAs to the project for
|
||||
T.38 testing. (http://www.grandstream.com)
|
||||
=== DEVELOPMENT SUPPORT ===
|
||||
I'd like to thank the following companies for helping fund development of
|
||||
Asterisk:
|
||||
|
||||
=== MISCELLANEOUS PATCHES ===
|
||||
Jim Dixon - Zapata Telephony and app_rpt
|
||||
http://www.zapatatelephony.org/app_rpt.html
|
||||
Pilosoft, Inc. - for supporting ADSI development in Asterisk
|
||||
|
||||
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; 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, SNMP agent
|
||||
support (res_snmp), 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, $[ ] parser upgrade, AEL2 parser upgrade
|
||||
|
||||
Claude Patry - bug fixes, feature enhancements, and bug marshalling
|
||||
cpatry@gmail.com
|
||||
|
||||
Miroslav Nachev, miro@space-comm.com COSMOS Software Enterprises, Ltd.
|
||||
- for Variable for No Answer Timeout for Attended Transfer
|
||||
|
||||
Slav Klenov & Vanheuverzwijn Joachim - development of the generic jitterbuffer
|
||||
Securax Ltd. info@securax.be
|
||||
|
||||
Roy Sigurd Karlsbakk - providing funding for generic jitterbuffer development
|
||||
roy@karlsbakk.net, Briiz Telecom AS
|
||||
|
||||
Voop A/S, Nuvio Inc, Inotel S.A and Foniris Telecom A/S - funding for rewrite of SIP transfers
|
||||
|
||||
Philippe Sultan - RADIUS CDR module
|
||||
INRIA, http://www.inria.fr/
|
||||
|
||||
John Martin, Aupix - Improved video support in the SIP channel
|
||||
|
||||
Steve Underwood - Provided T.38 pass through support.
|
||||
|
||||
George Konstantoulakis - Support for Greek in voicemail added by InAccess Networks (work funded by HOL, www.hol.gr) gkon@inaccessnetworks.com
|
||||
|
||||
Daniel Nylander - Support for Swedish and Norwegian languages in voicemail. http://www.danielnylander.se/
|
||||
|
||||
Stojan Sljivic - An option for maximum number of messsages per mailbox in voicemail. Also an issue with voicemail synchronization has been fixed. GDS Partners www.gdspartners.com . stojan.sljivic@gdspartners.com
|
||||
|
||||
Bartosz Supczinski - Support for Polish added by DIR (www.dir.pl) Bartosz.Supczinski@dir.pl
|
||||
|
||||
James Rothenberger - Support for IMAP storage integration added by OneBizTone LLC Work funded by University of Pennsylvania jar@onebiztone.com
|
||||
|
||||
Paul Cadach - Bringing chan_h323 up to date, bug fixes, and more!
|
||||
|
||||
=== OTHER CONTRIBUTIONS ===
|
||||
John Todd - Monkey sounds and associated teletorture prompt
|
||||
Michael Jerris - bug marshaling
|
||||
Leif Madsen, Jared Smith and Jim van Meggelen - the Asterisk book
|
||||
available under a Creative Commons License at http://www.asteriskdocs.org
|
||||
Brian M. Clapper - poll.c emulation
|
||||
This product includes software developed by Brian M. Clapper <bmc@clapper.org>
|
||||
|
||||
=== HOLD MUSIC ===
|
||||
Music provided by www.opsound.org
|
||||
|
||||
=== OTHER SOURCE CODE IN ASTERISK ===
|
||||
Asterisk uses libedit, the lightweight readline replacement from NetBSD.
|
||||
The cdr_radius module uses libradiusclient-ng, which is also from NetBSD.
|
||||
They are BSD-licensed and require the following statement:
|
||||
|
||||
This product includes software developed by the NetBSD
|
||||
Foundation, Inc. and its contributors.
|
||||
|
||||
Digium did not implement the codecs in Asterisk. Here is the copyright on the
|
||||
I did not implement the codecs in asterisk. Here is the copyright on the
|
||||
GSM source:
|
||||
|
||||
Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
|
||||
@@ -196,7 +37,7 @@ 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,
|
||||
|
||||
390
LICENSE
390
LICENSE
@@ -1,69 +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.
|
||||
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 more information on this policy, please read
|
||||
http://www.digium.com/en/company/profile/trademarkpolicy.php
|
||||
For example, if you distribute copies of such a program, whether
|
||||
gratis or for a fee, you must give the recipients all the rights that
|
||||
you have. You must make sure that they, too, receive or can get the
|
||||
source code. And you must show them these terms so they know their
|
||||
rights.
|
||||
|
||||
If you have any questions regarding our licensing policy, please
|
||||
contact us:
|
||||
We protect your rights with two steps: (1) copyright the software, and
|
||||
(2) offer you this license which gives you legal permission to copy,
|
||||
distribute and/or modify the software.
|
||||
|
||||
+1.877.546.8963 (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/misery.digium.com/6000 (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.
|
||||
|
||||
GNU GENERAL PUBLIC LICENSE
|
||||
TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
|
||||
|
||||
0. This License applies to any program or other work which contains
|
||||
a notice placed by the copyright holder saying it may be distributed
|
||||
under the terms of this General Public License. The "Program", below,
|
||||
refers to any such program or work, and a "work based on the Program"
|
||||
means either the Program or any derivative work under copyright law:
|
||||
that is to say, a work containing the Program or a portion of it,
|
||||
either verbatim or with modifications and/or translated into another
|
||||
language. (Hereinafter, translation is included without limitation in
|
||||
the term "modification".) Each licensee is addressed as "you".
|
||||
|
||||
Activities other than copying, distribution and modification are not
|
||||
covered by this License; they are outside its scope. The act of
|
||||
running the Program is not restricted, and the output from the Program
|
||||
is covered only if its contents constitute a work based on the
|
||||
Program (independent of having been made by running the Program).
|
||||
Whether that is true depends on what the Program does.
|
||||
|
||||
1. You may copy and distribute verbatim copies of the Program's
|
||||
source code as you receive it, in any medium, provided that you
|
||||
conspicuously and appropriately publish on each copy an appropriate
|
||||
copyright notice and disclaimer of warranty; keep intact all the
|
||||
notices that refer to this License and to the absence of any warranty;
|
||||
and give any other recipients of the Program a copy of this License
|
||||
along with the Program.
|
||||
|
||||
You may charge a fee for the physical act of transferring a copy, and
|
||||
you may at your option offer warranty protection in exchange for a fee.
|
||||
|
||||
2. You may modify your copy or copies of the Program or any portion
|
||||
of it, thus forming a work based on the Program, and copy and
|
||||
distribute such modifications or work under the terms of Section 1
|
||||
above, provided that you also meet all of these conditions:
|
||||
|
||||
a) You must cause the modified files to carry prominent notices
|
||||
stating that you changed the files and the date of any change.
|
||||
|
||||
b) You must cause any work that you distribute or publish, that in
|
||||
whole or in part contains or is derived from the Program or any
|
||||
part thereof, to be licensed as a whole at no charge to all third
|
||||
parties under the terms of this License.
|
||||
|
||||
c) If the modified program normally reads commands interactively
|
||||
when run, you must cause it, when started running for such
|
||||
interactive use in the most ordinary way, to print or display an
|
||||
announcement including an appropriate copyright notice and a
|
||||
notice that there is no warranty (or else, saying that you provide
|
||||
a warranty) and that users may redistribute the program under
|
||||
these conditions, and telling the user how to view a copy of this
|
||||
License. (Exception: if the Program itself is interactive but
|
||||
does not normally print such an announcement, your work based on
|
||||
the Program is not required to print an announcement.)
|
||||
|
||||
These requirements apply to the modified work as a whole. If
|
||||
identifiable sections of that work are not derived from the Program,
|
||||
and can be reasonably considered independent and separate works in
|
||||
themselves, then this License, and its terms, do not apply to those
|
||||
sections when you distribute them as separate works. But when you
|
||||
distribute the same sections as part of a whole which is a work based
|
||||
on the Program, the distribution of the whole must be on the terms of
|
||||
this License, whose permissions for other licensees extend to the
|
||||
entire whole, and thus to each and every part regardless of who wrote it.
|
||||
|
||||
Thus, it is not the intent of this section to claim rights or contest
|
||||
your rights to work written entirely by you; rather, the intent is to
|
||||
exercise the right to control the distribution of derivative or
|
||||
collective works based on the Program.
|
||||
|
||||
In addition, mere aggregation of another work not based on the Program
|
||||
with the Program (or with a work based on the Program) on a volume of
|
||||
a storage or distribution medium does not bring the other work under
|
||||
the scope of this License.
|
||||
|
||||
3. You may copy and distribute the Program (or a work based on it,
|
||||
under Section 2) in object code or executable form under the terms of
|
||||
Sections 1 and 2 above provided that you also do one of the following:
|
||||
|
||||
a) Accompany it with the complete corresponding machine-readable
|
||||
source code, which must be distributed under the terms of Sections
|
||||
1 and 2 above on a medium customarily used for software interchange; or,
|
||||
|
||||
b) Accompany it with a written offer, valid for at least three
|
||||
years, to give any third party, for a charge no more than your
|
||||
cost of physically performing source distribution, a complete
|
||||
machine-readable copy of the corresponding source code, to be
|
||||
distributed under the terms of Sections 1 and 2 above on a medium
|
||||
customarily used for software interchange; or,
|
||||
|
||||
c) Accompany it with the information you received as to the offer
|
||||
to distribute corresponding source code. (This alternative is
|
||||
allowed only for noncommercial distribution and only if you
|
||||
received the program in object code or executable form with such
|
||||
an offer, in accord with Subsection b above.)
|
||||
|
||||
The source code for a work means the preferred form of the work for
|
||||
making modifications to it. For an executable work, complete source
|
||||
code means all the source code for all modules it contains, plus any
|
||||
associated interface definition files, plus the scripts used to
|
||||
control compilation and installation of the executable. However, as a
|
||||
special exception, the source code distributed need not include
|
||||
anything that is normally distributed (in either source or binary
|
||||
form) with the major components (compiler, kernel, and so on) of the
|
||||
operating system on which the executable runs, unless that component
|
||||
itself accompanies the executable.
|
||||
|
||||
If distribution of executable or object code is made by offering
|
||||
access to copy from a designated place, then offering equivalent
|
||||
access to copy the source code from the same place counts as
|
||||
distribution of the source code, even though third parties are not
|
||||
compelled to copy the source along with the object code.
|
||||
|
||||
4. You may not copy, modify, sublicense, or distribute the Program
|
||||
except as expressly provided under this License. Any attempt
|
||||
otherwise to copy, modify, sublicense or distribute the Program is
|
||||
void, and will automatically terminate your rights under this License.
|
||||
However, parties who have received copies, or rights, from you under
|
||||
this License will not have their licenses terminated so long as such
|
||||
parties remain in full compliance.
|
||||
|
||||
5. You are not required to accept this License, since you have not
|
||||
signed it. However, nothing else grants you permission to modify or
|
||||
distribute the Program or its derivative works. These actions are
|
||||
prohibited by law if you do not accept this License. Therefore, by
|
||||
modifying or distributing the Program (or any work based on the
|
||||
Program), you indicate your acceptance of this License to do so, and
|
||||
all its terms and conditions for copying, distributing or modifying
|
||||
the Program or works based on it.
|
||||
|
||||
6. Each time you redistribute the Program (or any work based on the
|
||||
Program), the recipient automatically receives a license from the
|
||||
original licensor to copy, distribute or modify the Program subject to
|
||||
these terms and conditions. You may not impose any further
|
||||
restrictions on the recipients' exercise of the rights granted herein.
|
||||
You are not responsible for enforcing compliance by third parties to
|
||||
this License.
|
||||
|
||||
7. If, as a consequence of a court judgment or allegation of patent
|
||||
infringement or for any other reason (not limited to patent issues),
|
||||
conditions are imposed on you (whether by court order, agreement or
|
||||
otherwise) that contradict the conditions of this License, they do not
|
||||
excuse you from the conditions of this License. If you cannot
|
||||
distribute so as to satisfy simultaneously your obligations under this
|
||||
License and any other pertinent obligations, then as a consequence you
|
||||
may not distribute the Program at all. For example, if a patent
|
||||
license would not permit royalty-free redistribution of the Program by
|
||||
all those who receive copies directly or indirectly through you, then
|
||||
the only way you could satisfy both it and this License would be to
|
||||
refrain entirely from distribution of the Program.
|
||||
|
||||
If any portion of this section is held invalid or unenforceable under
|
||||
any particular circumstance, the balance of the section is intended to
|
||||
apply and the section as a whole is intended to apply in other
|
||||
circumstances.
|
||||
|
||||
It is not the purpose of this section to induce you to infringe any
|
||||
patents or other property right claims or to contest validity of any
|
||||
such claims; this section has the sole purpose of protecting the
|
||||
integrity of the free software distribution system, which is
|
||||
implemented by public license practices. Many people have made
|
||||
generous contributions to the wide range of software distributed
|
||||
through that system in reliance on consistent application of that
|
||||
system; it is up to the author/donor to decide if he or she is willing
|
||||
to distribute software through any other system and a licensee cannot
|
||||
impose that choice.
|
||||
|
||||
This section is intended to make thoroughly clear what is believed to
|
||||
be a consequence of the rest of this License.
|
||||
|
||||
8. If the distribution and/or use of the Program is restricted in
|
||||
certain countries either by patents or by copyrighted interfaces, the
|
||||
original copyright holder who places the Program under this License
|
||||
may add an explicit geographical distribution limitation excluding
|
||||
those countries, so that distribution is permitted only in or among
|
||||
countries not thus excluded. In such case, this License incorporates
|
||||
the limitation as if written in the body of this License.
|
||||
|
||||
9. The Free Software Foundation may publish revised and/or new versions
|
||||
of the General Public License from time to time. Such new versions will
|
||||
be similar in spirit to the present version, but may differ in detail to
|
||||
address new problems or concerns.
|
||||
|
||||
Each version is given a distinguishing version number. If the Program
|
||||
specifies a version number of this License which applies to it and "any
|
||||
later version", you have the option of following the terms and conditions
|
||||
either of that version or of any later version published by the Free
|
||||
Software Foundation. If the Program does not specify a version number of
|
||||
this License, you may choose any version ever published by the Free Software
|
||||
Foundation.
|
||||
|
||||
10. If you wish to incorporate parts of the Program into other free
|
||||
programs whose distribution conditions are different, write to the author
|
||||
to ask for permission. For software which is copyrighted by the Free
|
||||
Software Foundation, write to the Free Software Foundation; we sometimes
|
||||
make exceptions for this. Our decision will be guided by the two goals
|
||||
of preserving the free status of all derivatives of our free software and
|
||||
of promoting the sharing and reuse of software generally.
|
||||
|
||||
NO WARRANTY
|
||||
|
||||
11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY
|
||||
FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHEN
|
||||
OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES
|
||||
PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED
|
||||
OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
|
||||
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. THE ENTIRE RISK AS
|
||||
TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU. SHOULD THE
|
||||
PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING,
|
||||
REPAIR OR CORRECTION.
|
||||
|
||||
12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
|
||||
WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR
|
||||
REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES,
|
||||
INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING
|
||||
OUT OF THE USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED
|
||||
TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY
|
||||
YOU OR THIRD PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER
|
||||
PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE
|
||||
POSSIBILITY OF SUCH DAMAGES.
|
||||
|
||||
END OF TERMS AND CONDITIONS
|
||||
|
||||
How to Apply These Terms to Your New Programs
|
||||
|
||||
If you develop a new program, and you want it to be of the greatest
|
||||
possible use to the public, the best way to achieve this is to make it
|
||||
free software which everyone can redistribute and change under these terms.
|
||||
|
||||
To do so, attach the following notices to the program. It is safest
|
||||
to attach them to the start of each source file to most effectively
|
||||
convey the exclusion of warranty; and each file should have at least
|
||||
the "copyright" line and a pointer to where the full notice is found.
|
||||
|
||||
<one line to give the program's name and a brief idea of what it does.>
|
||||
Copyright (C) 19yy <name of author>
|
||||
|
||||
This program is free software; you can redistribute it and/or modify
|
||||
it under the terms of the GNU General Public License as published by
|
||||
the Free Software Foundation; either version 2 of the License, or
|
||||
(at your option) any later version.
|
||||
|
||||
This program is distributed in the hope that it will be useful,
|
||||
but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
||||
GNU General Public License for more details.
|
||||
|
||||
You should have received a copy of the GNU General Public License
|
||||
along with this program; if not, write to the Free Software
|
||||
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
|
||||
|
||||
Also add information on how to contact you by electronic and paper mail.
|
||||
|
||||
If the program is interactive, make it output a short notice like this
|
||||
when it starts in an interactive mode:
|
||||
|
||||
Gnomovision version 69, Copyright (C) 19yy name of author
|
||||
Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
|
||||
This is free software, and you are welcome to redistribute it
|
||||
under certain conditions; type `show c' for details.
|
||||
|
||||
The hypothetical commands `show w' and `show c' should show the appropriate
|
||||
parts of the General Public License. Of course, the commands you use may
|
||||
be called something other than `show w' and `show c'; they could even be
|
||||
mouse-clicks or menu items--whatever suits your program.
|
||||
|
||||
You should also get your employer (if you work as a programmer) or your
|
||||
school, if any, to sign a "copyright disclaimer" for the program, if
|
||||
necessary. Here is a sample; alter the names:
|
||||
|
||||
Yoyodyne, Inc., hereby disclaims all copyright interest in the program
|
||||
`Gnomovision' (which makes passes at compilers) written by James Hacker.
|
||||
|
||||
<signature of Ty Coon>, 1 April 1989
|
||||
Ty Coon, President of Vice
|
||||
|
||||
This General Public License does not permit incorporating your program into
|
||||
proprietary programs. If your program is a subroutine library, you may
|
||||
consider it more useful to permit linking proprietary applications with the
|
||||
library. If this is what you want to do, use the GNU Library General
|
||||
Public License instead of this License.
|
||||
|
||||
735
Makefile
735
Makefile
@@ -3,475 +3,110 @@
|
||||
#
|
||||
# Top level Makefile
|
||||
#
|
||||
# Copyright (C) 1999-2006, Digium, Inc.
|
||||
# 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
|
||||
#
|
||||
|
||||
# All Makefiles use the following variables:
|
||||
#
|
||||
# ASTCFLAGS - compiler options
|
||||
# ASTLDFLAGS - linker flags (not libraries)
|
||||
# AST_LIBS - libraries to build binaries XXX
|
||||
# LIBS - additional libraries, at top-level for all links,
|
||||
# on a single object just for that object
|
||||
# SOLINK - linker flags used only for creating shared objects (.so files),
|
||||
# used for all .so links
|
||||
#
|
||||
# Default values fo ASTCFLAGS and ASTLDFLAGS can be specified in the
|
||||
# environment when running make, as follows:
|
||||
#
|
||||
# $ ASTCFLAGS="-Werror" make
|
||||
|
||||
export ASTTOPDIR
|
||||
export ASTERISKVERSION
|
||||
export ASTERISKVERSIONNUM
|
||||
export INSTALL_PATH
|
||||
export ASTETCDIR
|
||||
export ASTVARRUNDIR
|
||||
export MODULES_DIR
|
||||
export ASTSPOOLDIR
|
||||
export ASTVARLIBDIR
|
||||
export ASTDATADIR
|
||||
export ASTLOGDIR
|
||||
export ASTLIBDIR
|
||||
export ASTMANDIR
|
||||
export ASTHEADERDIR
|
||||
export ASTBINDIR
|
||||
export ASTSBINDIR
|
||||
export AGI_DIR
|
||||
export ASTCONFPATH
|
||||
export NOISY_BUILD
|
||||
export MENUSELECT_CFLAGS
|
||||
export CC
|
||||
export CXX
|
||||
export AR
|
||||
export RANLIB
|
||||
export HOST_CC
|
||||
export STATIC_BUILD
|
||||
export INSTALL
|
||||
export DESTDIR
|
||||
export PROC
|
||||
export SOLINK
|
||||
export STRIP
|
||||
export DOWNLOAD
|
||||
export OSARCH
|
||||
export CURSES_DIR
|
||||
export NCURSES_DIR
|
||||
export TERMCAP_DIR
|
||||
export TINFO_DIR
|
||||
export GTK2_LIB
|
||||
export GTK2_INCLUDE
|
||||
.EXPORT_ALL_VARIABLES:
|
||||
|
||||
# even though we could use '-include makeopts' here, use a wildcard
|
||||
# lookup anyway, so that make won't try to build makeopts if it doesn't
|
||||
# exist (other rules will force it to be built if needed)
|
||||
ifneq ($(wildcard makeopts),)
|
||||
include makeopts
|
||||
endif
|
||||
INSTALL_PREFIX=
|
||||
|
||||
#Uncomment this to see all build commands instead of 'quiet' output
|
||||
#NOISY_BUILD=yes
|
||||
MODULES_DIR=$(INSTALL_PREFIX)/usr/lib/asterisk/modules
|
||||
AGI_DIR=$(INSTALL_PREFIX)/var/lib/asterisk/agi-bin
|
||||
|
||||
# Create OPTIONS variable
|
||||
OPTIONS=
|
||||
# Pentium Pro Optimize
|
||||
#PROC=i686
|
||||
# Pentium Optimize
|
||||
PROC=i586
|
||||
|
||||
ASTTOPDIR:=$(shell pwd)
|
||||
DEBUG=-g #-pg
|
||||
INCLUDE=-Iinclude -I../include
|
||||
CFLAGS=-pipe -Wall -Wmissing-prototypes -Wmissing-declarations -O6 $(DEBUG) $(INCLUDE) -D_REENTRANT
|
||||
#CFLAGS+=-Werror
|
||||
CFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
|
||||
ASTERISKVERSION=$(shell if [ -f .version ]; then cat .version; fi)
|
||||
RPMVERSION=$(shell sed 's/[-\/:]/_/g' .version)
|
||||
CFLAGS+=-DASTERISK_VERSION=\"$(ASTERISKVERSION)\"
|
||||
# Optional debugging parameters
|
||||
CFLAGS+= -DDO_CRASH -DDEBUG_THREADS
|
||||
# Uncomment next one to enable ast_frame tracing (for debugging)
|
||||
#CLFAGS+= -DTRACE_FRAMES
|
||||
CFLAGS+=# -fomit-frame-pointer
|
||||
SUBDIRS=res channels pbx apps codecs formats agi cdr
|
||||
LIBS=-ldl -lpthread -lreadline -lncurses -lm
|
||||
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 \
|
||||
ulaw.o alaw.o callerid.o fskmodem.o image.o app.o \
|
||||
cdr.o tdd.o asterisk.o
|
||||
CC=gcc
|
||||
INSTALL=install
|
||||
|
||||
# Overwite config files on "make samples"
|
||||
OVERWRITE=y
|
||||
|
||||
# Include debug and macro symbols in the executables (-g) and profiling info (-pg)
|
||||
DEBUG=-g3
|
||||
|
||||
# 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
|
||||
# !!! Watch out, put no spaces or comments after the value !!!
|
||||
#DESTDIR?=/tmp/asterisk
|
||||
|
||||
# Define standard directories for various platforms
|
||||
# These apply if they are not redefined in asterisk.conf
|
||||
ifeq ($(OSARCH),SunOS)
|
||||
ASTETCDIR=/var/etc/asterisk
|
||||
ASTLIBDIR=/opt/asterisk/lib
|
||||
ASTVARLIBDIR=/var/opt/asterisk
|
||||
ASTSPOOLDIR=/var/spool/asterisk
|
||||
ASTLOGDIR=/var/log/asterisk
|
||||
ASTHEADERDIR=/opt/asterisk/include
|
||||
ASTBINDIR=/opt/asterisk/bin
|
||||
ASTSBINDIR=/opt/asterisk/sbin
|
||||
ASTVARRUNDIR=/var/run/asterisk
|
||||
ASTMANDIR=/opt/asterisk/man
|
||||
else
|
||||
ASTETCDIR=$(sysconfdir)/asterisk
|
||||
ASTLIBDIR=$(libdir)/asterisk
|
||||
ASTHEADERDIR=$(includedir)/asterisk
|
||||
ASTBINDIR=$(bindir)
|
||||
ASTSBINDIR=$(sbindir)
|
||||
ASTSPOOLDIR=$(localstatedir)/spool/asterisk
|
||||
ASTLOGDIR=$(localstatedir)/log/asterisk
|
||||
ASTVARRUNDIR=$(localstatedir)/run
|
||||
ASTMANDIR=$(mandir)
|
||||
ifeq ($(OSARCH),FreeBSD)
|
||||
ASTVARLIBDIR=$(prefix)/share/asterisk
|
||||
else
|
||||
ASTVARLIBDIR=$(localstatedir)/lib/asterisk
|
||||
endif
|
||||
endif
|
||||
ifeq ($(ASTDATADIR),)
|
||||
ASTDATADIR:=$(ASTVARLIBDIR)
|
||||
endif
|
||||
|
||||
# Asterisk.conf is located in ASTETCDIR or by using the -C flag
|
||||
# when starting Asterisk
|
||||
ASTCONFPATH=$(ASTETCDIR)/asterisk.conf
|
||||
MODULES_DIR=$(ASTLIBDIR)/modules
|
||||
AGI_DIR=$(ASTDATADIR)/agi-bin
|
||||
|
||||
# If you use Apache, you may determine by a grep 'DocumentRoot' of your httpd.conf file
|
||||
HTTP_DOCSDIR=/var/www/html
|
||||
# Determine by a grep 'ScriptAlias' of your Apache httpd.conf file
|
||||
HTTP_CGIDIR=/var/www/cgi-bin
|
||||
|
||||
# Uncomment this to use the older DSP routines
|
||||
#ASTCFLAGS+=-DOLD_DSP_ROUTINES
|
||||
|
||||
# If the file .asterisk.makeopts is present in your home directory, you can
|
||||
# include all of your favorite menuselect options so that every time you download
|
||||
# a new version of Asterisk, you don't have to run menuselect to set them.
|
||||
# The file /etc/asterisk.makeopts will also be included but can be overridden
|
||||
# by the file in your home directory.
|
||||
|
||||
GLOBAL_MAKEOPTS=$(wildcard /etc/asterisk.makeopts)
|
||||
USER_MAKEOPTS=$(wildcard ~/.asterisk.makeopts)
|
||||
|
||||
MOD_SUBDIR_CFLAGS=-I$(ASTTOPDIR)/include
|
||||
OTHER_SUBDIR_CFLAGS=-I$(ASTTOPDIR)/include
|
||||
|
||||
ifeq ($(OSARCH),linux-gnu)
|
||||
ifeq ($(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
|
||||
endif
|
||||
|
||||
ASTCFLAGS+=-pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(DEBUG)
|
||||
|
||||
ASTCFLAGS+=-include $(ASTTOPDIR)/include/asterisk/autoconfig.h
|
||||
|
||||
ifeq ($(AST_DEVMODE),yes)
|
||||
ASTCFLAGS+=-Werror -Wunused
|
||||
endif
|
||||
|
||||
ifneq ($(findstring BSD,$(OSARCH)),)
|
||||
ASTCFLAGS+=-I/usr/local/include
|
||||
ASTLDFLAGS+=-L/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
|
||||
|
||||
ifeq ($(OSARCH),FreeBSD)
|
||||
# -V is understood by BSD Make, not by GNU make.
|
||||
BSDVERSION=$(shell make -V OSVERSION -f /usr/share/mk/bsd.port.subdir.mk)
|
||||
ASTCFLAGS+=$(shell if test $(BSDVERSION) -lt 500016 ; then echo "-D_THREAD_SAFE"; fi)
|
||||
AST_LIBS+=$(shell if test $(BSDVERSION) -lt 502102 ; then echo "-lc_r"; else echo "-pthread"; fi)
|
||||
endif
|
||||
|
||||
ifeq ($(OSARCH),NetBSD)
|
||||
ASTCFLAGS+=-pthread -I/usr/pkg/include
|
||||
endif
|
||||
|
||||
ifeq ($(OSARCH),OpenBSD)
|
||||
ASTCFLAGS+=-pthread
|
||||
endif
|
||||
|
||||
ifeq ($(OSARCH),SunOS)
|
||||
ASTCFLAGS+=-Wcast-align -DSOLARIS -I../include/solaris-compat -I/opt/ssl/include -I/usr/local/ssl/include
|
||||
endif
|
||||
|
||||
ASTERISKVERSION:=$(shell build_tools/make_version .)
|
||||
|
||||
ifneq ($(wildcard .version),)
|
||||
ASTERISKVERSIONNUM:=$(shell awk -F. '{printf "%01d%02d%02d", $$1, $$2, $$3}' .version)
|
||||
RPMVERSION:=$(shell sed 's/[-\/:]/_/g' .version)
|
||||
else
|
||||
RPMVERSION=unknown
|
||||
endif
|
||||
|
||||
ifneq ($(wildcard .svn),)
|
||||
ASTERISKVERSIONNUM=999999
|
||||
endif
|
||||
|
||||
ASTCFLAGS+=$(MALLOC_DEBUG)$(BUSYDETECT)$(OPTIONS)
|
||||
|
||||
MOD_SUBDIRS:=res channels pbx apps codecs formats cdr funcs main
|
||||
OTHER_SUBDIRS:=utils agi
|
||||
SUBDIRS:=$(OTHER_SUBDIRS) $(MOD_SUBDIRS)
|
||||
SUBDIRS_INSTALL:=$(SUBDIRS:%=%-install)
|
||||
SUBDIRS_CLEAN:=$(SUBDIRS:%=%-clean)
|
||||
SUBDIRS_UNINSTALL:=$(SUBDIRS:%=%-uninstall)
|
||||
MOD_SUBDIRS_EMBED_LDSCRIPT:=$(MOD_SUBDIRS:%=%-embed-ldscript)
|
||||
MOD_SUBDIRS_EMBED_LDFLAGS:=$(MOD_SUBDIRS:%=%-embed-ldflags)
|
||||
MOD_SUBDIRS_EMBED_LIBS:=$(MOD_SUBDIRS:%=%-embed-libs)
|
||||
|
||||
ifneq ($(findstring darwin,$(OSARCH)),)
|
||||
ASTCFLAGS+=-D__Darwin__
|
||||
AUDIO_LIBS=-framework CoreAudio
|
||||
SOLINK=-dynamic -bundle -undefined suppress -force_flat_namespace
|
||||
else
|
||||
# These are used for all but Darwin
|
||||
SOLINK=-shared -Xlinker -x
|
||||
ifneq ($(findstring BSD,$(OSARCH)),)
|
||||
LDFLAGS+=-L/usr/local/lib
|
||||
endif
|
||||
endif
|
||||
|
||||
ifeq ($(OSARCH),SunOS)
|
||||
SOLINK=-shared -fpic -L/usr/local/ssl/lib
|
||||
endif
|
||||
|
||||
# This is used when generating the doxygen documentation
|
||||
ifneq ($(DOT),:)
|
||||
HAVEDOT=yes
|
||||
else
|
||||
HAVEDOT=no
|
||||
endif
|
||||
|
||||
all: _all
|
||||
_all: all
|
||||
@echo " +--------- Asterisk Build Complete ---------+"
|
||||
@echo " + Asterisk has successfully been built, and +"
|
||||
@echo " + can be installed by running: +"
|
||||
@echo " + Asterisk has successfully been built, but +"
|
||||
@echo " + cannot be run before being installed by +"
|
||||
@echo " + running: +"
|
||||
@echo " + +"
|
||||
@echo " + $(MAKE) install +"
|
||||
@echo " + make install +"
|
||||
@echo " +-------------------------------------------+"
|
||||
|
||||
_all: cleantest $(SUBDIRS)
|
||||
all: asterisk subdirs
|
||||
|
||||
makeopts: configure
|
||||
@echo "****"
|
||||
@echo "**** The configure script must be executed before running '$(MAKE)'."
|
||||
@echo "**** Please run \"./configure\"."
|
||||
@echo "****"
|
||||
@exit 1
|
||||
_version:
|
||||
if [ -d CVS ] && ! [ -f .version ]; then echo "CVS-`date +"%D-%T"`" > .version; fi
|
||||
|
||||
menuselect.makeopts: menuselect/menuselect menuselect-tree
|
||||
menuselect/menuselect --check-deps $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS) menuselect.makeopts
|
||||
.version: _version
|
||||
|
||||
$(MOD_SUBDIRS_EMBED_LDSCRIPT):
|
||||
@echo "EMBED_LDSCRIPTS+="`$(MAKE) --quiet --no-print-directory -C $(@:-embed-ldscript=) SUBDIR=$(@:-embed-ldscript=) __embed_ldscript` >> makeopts.embed_rules
|
||||
build.h:
|
||||
./make_build_h
|
||||
|
||||
$(MOD_SUBDIRS_EMBED_LDFLAGS):
|
||||
@echo "EMBED_LDFLAGS+="`$(MAKE) --quiet --no-print-directory -C $(@:-embed-ldflags=) SUBDIR=$(@:-embed-ldflags=) __embed_ldflags` >> makeopts.embed_rules
|
||||
asterisk: .version build.h $(OBJS)
|
||||
gcc -o asterisk -rdynamic $(OBJS) $(LIBS)
|
||||
|
||||
$(MOD_SUBDIRS_EMBED_LIBS):
|
||||
@echo "EMBED_LIBS+="`$(MAKE) --quiet --no-print-directory -C $(@:-embed-libs=) SUBDIR=$(@:-embed-libs=) __embed_libs` >> makeopts.embed_rules
|
||||
subdirs:
|
||||
for x in $(SUBDIRS); do $(MAKE) -C $$x || exit 1 ; done
|
||||
|
||||
makeopts.embed_rules: menuselect.makeopts
|
||||
@echo "Generating embedded module rules ..."
|
||||
@rm -f $@
|
||||
@$(MAKE) --no-print-directory $(MOD_SUBDIRS_EMBED_LDSCRIPT)
|
||||
@$(MAKE) --no-print-directory $(MOD_SUBDIRS_EMBED_LDFLAGS)
|
||||
@$(MAKE) --no-print-directory $(MOD_SUBDIRS_EMBED_LIBS)
|
||||
clean:
|
||||
for x in $(SUBDIRS); do $(MAKE) -C $$x clean || exit 1 ; done
|
||||
rm -f *.o *.so asterisk
|
||||
rm -f build.h
|
||||
|
||||
$(SUBDIRS): include/asterisk/version.h include/asterisk/buildopts.h defaults.h makeopts.embed_rules
|
||||
|
||||
# ensure that all module subdirectories are processed before 'main' during
|
||||
# a parallel build, since if there are modules selected to be embedded the
|
||||
# directories containing them must be completed before the main Asterisk
|
||||
# binary can be built
|
||||
main: $(filter-out main,$(MOD_SUBDIRS))
|
||||
|
||||
$(MOD_SUBDIRS):
|
||||
@ASTCFLAGS="$(MOD_SUBDIR_CFLAGS) $(ASTCFLAGS)" ASTLDFLAGS="$(ASTLDFLAGS)" AST_LIBS="$(AST_LIBS)" $(MAKE) --no-print-directory --no-builtin-rules -C $@ SUBDIR=$@ all
|
||||
|
||||
$(OTHER_SUBDIRS):
|
||||
@ASTCFLAGS="$(OTHER_SUBDIR_CFLAGS) $(ASTCFLAGS)" ASTLDFLAGS="$(ASTLDFLAGS)" AUDIO_LIBS="$(AUDIO_LIBS)" $(MAKE) --no-print-directory --no-builtin-rules -C $@ SUBDIR=$@ all
|
||||
|
||||
defaults.h: makeopts
|
||||
@build_tools/make_defaults_h > $@.tmp
|
||||
@if cmp -s $@.tmp $@ ; then : ; else \
|
||||
mv $@.tmp $@ ; \
|
||||
fi
|
||||
@rm -f $@.tmp
|
||||
|
||||
include/asterisk/version.h:
|
||||
@build_tools/make_version_h > $@.tmp
|
||||
@if cmp -s $@.tmp $@ ; then : ; else \
|
||||
mv $@.tmp $@ ; \
|
||||
fi
|
||||
@rm -f $@.tmp
|
||||
|
||||
include/asterisk/buildopts.h: menuselect.makeopts
|
||||
@build_tools/make_buildopts_h > $@.tmp
|
||||
@if cmp -s $@.tmp $@ ; then : ; else \
|
||||
mv $@.tmp $@ ; \
|
||||
fi
|
||||
@rm -f $@.tmp
|
||||
|
||||
$(SUBDIRS_CLEAN):
|
||||
@$(MAKE) --no-print-directory -C $(@:-clean=) clean
|
||||
|
||||
clean: $(SUBDIRS_CLEAN)
|
||||
rm -f defaults.h
|
||||
rm -f include/asterisk/build.h
|
||||
rm -f include/asterisk/version.h
|
||||
@$(MAKE) -C menuselect clean
|
||||
cp -f .cleancount .lastclean
|
||||
|
||||
dist-clean: distclean
|
||||
|
||||
distclean: clean
|
||||
@$(MAKE) -C menuselect dist-clean
|
||||
@$(MAKE) -C sounds dist-clean
|
||||
rm -f menuselect.makeopts makeopts menuselect-tree menuselect.makedeps
|
||||
rm -f makeopts.embed_rules
|
||||
rm -f config.log config.status
|
||||
rm -rf autom4te.cache
|
||||
rm -f include/asterisk/autoconfig.h
|
||||
rm -f include/asterisk/buildopts.h
|
||||
rm -rf doc/api
|
||||
rm -f build_tools/menuselect-deps
|
||||
|
||||
datafiles: _all
|
||||
if [ x`$(ID) -un` = xroot ]; then CFLAGS="$(ASTCFLAGS)" sh build_tools/mkpkgconfig $(DESTDIR)/usr/lib/pkgconfig; fi
|
||||
# Should static HTTP be installed during make samples or even with its own target ala
|
||||
# webvoicemail? There are portions here that *could* be customized but might also be
|
||||
# improved a lot. I'll put it here for now.
|
||||
mkdir -p $(DESTDIR)$(ASTDATADIR)/static-http
|
||||
for x in static-http/*; do \
|
||||
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTDATADIR)/static-http ; \
|
||||
datafiles: all
|
||||
mkdir -p $(INSTALL_PREFIX)/var/lib/asterisk/sounds/digits
|
||||
for x in sounds/digits/*; do \
|
||||
install $$x $(INSTALL_PREFIX)/var/lib/asterisk/sounds/digits ; \
|
||||
done
|
||||
mkdir -p $(DESTDIR)$(ASTDATADIR)/images
|
||||
for x in sounds/vm-* sounds/transfer* sounds/pbx-* sounds/ss-* sounds/beep* sounds/dir-*; do \
|
||||
install $$x $(INSTALL_PREFIX)/var/lib/asterisk/sounds ; \
|
||||
done
|
||||
mkdir -p $(INSTALL_PREFIX)/var/lib/asterisk/images
|
||||
for x in images/*.jpg; do \
|
||||
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTDATADIR)/images ; \
|
||||
install $$x $(INSTALL_PREFIX)/var/lib/asterisk/images ; \
|
||||
done
|
||||
mkdir -p $(DESTDIR)$(AGI_DIR)
|
||||
$(MAKE) -C sounds install
|
||||
mkdir -p $(AGI_DIR)
|
||||
|
||||
update:
|
||||
@if [ -d .svn ]; then \
|
||||
echo "Updating from Subversion..." ; \
|
||||
svn update | tee update.out; \
|
||||
rm -f .version; \
|
||||
if [ `grep -c ^C update.out` -gt 0 ]; then \
|
||||
echo ; echo "The following files have conflicts:" ; \
|
||||
grep ^C update.out | cut -b4- ; \
|
||||
fi ; \
|
||||
rm -f update.out; \
|
||||
else \
|
||||
echo "Not under version control"; \
|
||||
fi
|
||||
|
||||
NEWHEADERS=$(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
|
||||
$(INSTALL) -m 755 main/asterisk $(DESTDIR)$(ASTSBINDIR)/
|
||||
$(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 $(DESTDIR)$(includedir)
|
||||
$(INSTALL) -m 644 include/asterisk/*.h $(DESTDIR)$(ASTHEADERDIR)
|
||||
if [ -n "$(OLDHEADERS)" ]; then \
|
||||
rm -f $(addprefix $(DESTDIR)$(ASTHEADERDIR)/,$(OLDHEADERS)) ;\
|
||||
fi
|
||||
mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-csv
|
||||
mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-custom
|
||||
mkdir -p $(DESTDIR)$(ASTDATADIR)/keys
|
||||
mkdir -p $(DESTDIR)$(ASTDATADIR)/firmware
|
||||
mkdir -p $(DESTDIR)$(ASTDATADIR)/firmware/iax
|
||||
mkdir -p $(DESTDIR)$(ASTMANDIR)/man8
|
||||
$(INSTALL) -m 644 keys/iaxtel.pub $(DESTDIR)$(ASTDATADIR)/keys
|
||||
$(INSTALL) -m 644 keys/freeworlddialup.pub $(DESTDIR)$(ASTDATADIR)/keys
|
||||
$(INSTALL) -m 644 doc/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 [ -f contrib/firmware/iax/iaxy.bin ] ; then \
|
||||
$(INSTALL) -m 644 contrib/firmware/iax/iaxy.bin $(DESTDIR)$(ASTDATADIR)/firmware/iax/iaxy.bin; \
|
||||
fi
|
||||
|
||||
$(SUBDIRS_INSTALL):
|
||||
@DESTDIR="$(DESTDIR)" ASTSBINDIR="$(ASTSBINDIR)" $(MAKE) -C $(@:-install=) install
|
||||
|
||||
NEWMODS=$(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 " $$f" ;\
|
||||
done ;\
|
||||
echo "" ;\
|
||||
echo " WARNING WARNING WARNING" ;\
|
||||
fi
|
||||
|
||||
install: datafiles bininstall $(SUBDIRS_INSTALL)
|
||||
@if [ -x /usr/sbin/asterisk-post-install ]; then \
|
||||
/usr/sbin/asterisk-post-install $(DESTDIR) . ; \
|
||||
fi
|
||||
install: all datafiles
|
||||
mkdir -p $(MODULES_DIR)
|
||||
mkdir -p $(INSTALL_PREFIX)/usr/sbin
|
||||
install -m 755 asterisk $(INSTALL_PREFIX)/usr/sbin/
|
||||
install -m 755 astgenkey $(INSTALL_PREFIX)/usr/sbin/
|
||||
for x in $(SUBDIRS); do $(MAKE) -C $$x install || exit 1 ; done
|
||||
install -d $(INSTALL_PREFIX)/usr/include/asterisk
|
||||
install include/asterisk/*.h $(INSTALL_PREFIX)/usr/include/asterisk
|
||||
rm -f $(INSTALL_PREFIX)/var/lib/asterisk/sounds/vm
|
||||
mkdir -p $(INSTALL_PREFIX)/var/spool/asterisk/vm
|
||||
rm -f $(INSTALL_PREFIX)/usr/lib/asterisk/modules/chan_ixj.so
|
||||
rm -f $(INSTALL_PREFIX)/usr/lib/asterisk/modules/chan_tor.so
|
||||
mkdir -p $(INSTALL_PREFIX)/var/lib/asterisk/sounds
|
||||
mkdir -p $(INSTALL_PREFIX)/var/log/asterisk/cdr-csv
|
||||
mkdir -p $(INSTALL_PREFIX)/var/lib/asterisk/keys
|
||||
install -m 644 keys/iaxtel.pub $(INSTALL_PREFIX)/var/lib/asterisk/keys
|
||||
( cd $(INSTALL_PREFIX)/var/lib/asterisk/sounds ; ln -s ../../../spool/asterisk/vm . )
|
||||
@echo " +---- Asterisk Installation Complete -------+"
|
||||
@echo " + +"
|
||||
@echo " + YOU MUST READ THE SECURITY DOCUMENT +"
|
||||
@@ -481,210 +116,58 @@ install: datafiles bininstall $(SUBDIRS_INSTALL)
|
||||
@echo " + configuration files (overwriting any +"
|
||||
@echo " + existing config files), run: +"
|
||||
@echo " + +"
|
||||
@echo " + $(MAKE) samples +"
|
||||
@echo " + make samples +"
|
||||
@echo " + +"
|
||||
@echo " +----------------- or ---------------------+"
|
||||
@echo " + +"
|
||||
@echo " + You can go ahead and install the asterisk +"
|
||||
@echo " + program documentation now or later run: +"
|
||||
@echo " + +"
|
||||
@echo " + $(MAKE) progdocs +"
|
||||
@echo " + make progdocs +"
|
||||
@echo " + +"
|
||||
@echo " + **Note** This requires that you have +"
|
||||
@echo " + doxygen installed on your local system +"
|
||||
@echo " +-------------------------------------------+"
|
||||
@$(MAKE) -s oldmodcheck
|
||||
|
||||
upgrade: bininstall
|
||||
|
||||
adsi:
|
||||
mkdir -p $(DESTDIR)$(ASTETCDIR)
|
||||
for x in configs/*.adsi; do \
|
||||
if [ ! -f $(DESTDIR)$(ASTETCDIR)/$$x ]; then \
|
||||
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTETCDIR)/`$(BASENAME) $$x` ; \
|
||||
fi ; \
|
||||
done
|
||||
|
||||
samples: adsi
|
||||
mkdir -p $(DESTDIR)$(ASTETCDIR)
|
||||
samples: all datafiles
|
||||
mkdir -p $(INSTALL_PREFIX)/etc/asterisk
|
||||
for x in configs/*.sample; do \
|
||||
if [ -f $(DESTDIR)$(ASTETCDIR)/`$(BASENAME) $$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 ;\
|
||||
if [ -f $(INSTALL_PREFIX)/etc/asterisk/`basename $$x .sample` ]; then \
|
||||
mv -f $(INSTALL_PREFIX)/etc/asterisk/`basename $$x .sample` $(INSTALL_PREFIX)/etc/asterisk/`basename $$x .sample`.old ; \
|
||||
fi ; \
|
||||
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTETCDIR)/`$(BASENAME) $$x .sample` ;\
|
||||
install $$x $(INSTALL_PREFIX)/etc/asterisk/`basename $$x .sample` ;\
|
||||
done
|
||||
if [ "$(OVERWRITE)" = "y" ] || [ ! -f $(DESTDIR)$(ASTCONFPATH) ]; then \
|
||||
( \
|
||||
echo "[directories]" ; \
|
||||
echo "astetcdir => $(ASTETCDIR)" ; \
|
||||
echo "astmoddir => $(MODULES_DIR)" ; \
|
||||
echo "astvarlibdir => $(ASTVARLIBDIR)" ; \
|
||||
echo "astdatadir => $(ASTDATADIR)" ; \
|
||||
echo "astagidir => $(AGI_DIR)" ; \
|
||||
echo "astspooldir => $(ASTSPOOLDIR)" ; \
|
||||
echo "astrundir => $(ASTVARRUNDIR)" ; \
|
||||
echo "astlogdir => $(ASTLOGDIR)" ; \
|
||||
echo "" ; \
|
||||
echo ";[options]" ; \
|
||||
echo ";internal_timing = yes" ; \
|
||||
echo ";systemname = my_system_name ; prefix uniqueid with a system name for global uniqueness issues" ; \
|
||||
echo "; Changing the following lines may compromise your security." ; \
|
||||
echo ";[files]" ; \
|
||||
echo ";astctlpermissions = 0660" ; \
|
||||
echo ";astctlowner = root" ; \
|
||||
echo ";astctlgroup = apache" ; \
|
||||
echo ";astctl = asterisk.ctl" ; \
|
||||
) > $(DESTDIR)$(ASTCONFPATH) ; \
|
||||
else \
|
||||
echo "Skipping asterisk.conf creation"; \
|
||||
fi
|
||||
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/INBOX
|
||||
build_tools/make_sample_voicemail $(DESTDIR)/$(ASTDATADIR) $(DESTDIR)/$(ASTSPOOLDIR)
|
||||
|
||||
webvmail:
|
||||
@[ -d $(DESTDIR)$(HTTP_DOCSDIR)/ ] || ( printf "http docs directory not found.\nUpdate assignment of variable HTTP_DOCSDIR in Makefile!\n" && exit 1 )
|
||||
@[ -d $(DESTDIR)$(HTTP_CGIDIR) ] || ( printf "cgi-bin directory not found.\nUpdate assignment of variable HTTP_CGIDIR in Makefile!\n" && exit 1 )
|
||||
$(INSTALL) -m 4755 -o root -g root contrib/scripts/vmail.cgi $(DESTDIR)$(HTTP_CGIDIR)/vmail.cgi
|
||||
mkdir -p $(DESTDIR)$(HTTP_DOCSDIR)/_asterisk
|
||||
for x in images/*.gif; do \
|
||||
$(INSTALL) -m 644 $$x $(DESTDIR)$(HTTP_DOCSDIR)/_asterisk/; \
|
||||
for x in sounds/demo-*; do \
|
||||
install $$x $(INSTALL_PREFIX)/var/lib/asterisk/sounds; \
|
||||
done
|
||||
mkdir -p $(INSTALL_PREFIX)/var/spool/asterisk/vm/1234/INBOX
|
||||
:> $(INSTALL_PREFIX)/var/lib/asterisk/sounds/vm/1234/unavail.gsm
|
||||
for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isunavail; do \
|
||||
cat $(INSTALL_PREFIX)/var/lib/asterisk/sounds/$$x.gsm >> $(INSTALL_PREFIX)/var/lib/asterisk/sounds/vm/1234/unavail.gsm ; \
|
||||
done
|
||||
:> $(INSTALL_PREFIX)/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 $(INSTALL_PREFIX)/var/lib/asterisk/sounds/$$x.gsm >> $(INSTALL_PREFIX)/var/lib/asterisk/sounds/vm/1234/busy.gsm ; \
|
||||
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 " + +"
|
||||
@echo " +-------------------------------------------+"
|
||||
|
||||
spec:
|
||||
sed "s/^Version:.*/Version: $(RPMVERSION)/g" redhat/asterisk.spec > asterisk.spec ; \
|
||||
mailbox:
|
||||
./addmailbox
|
||||
|
||||
|
||||
rpm: __rpm
|
||||
|
||||
__rpm: include/asterisk/version.h include/asterisk/buildopts.h spec
|
||||
__rpm: _version
|
||||
rm -rf /tmp/asterisk ; \
|
||||
mkdir -p /tmp/asterisk/redhat/RPMS/i386 ; \
|
||||
$(MAKE) DESTDIR=/tmp/asterisk install ; \
|
||||
$(MAKE) DESTDIR=/tmp/asterisk samples ; \
|
||||
make INSTALL_PREFIX=/tmp/asterisk install ; \
|
||||
make INSTALL_PREFIX=/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
|
||||
cp -f redhat/asterisk /tmp/asterisk/etc/rc.d/init.d/ ; \
|
||||
cp -f redhat/rpmrc /tmp/asterisk/ ; \
|
||||
cp -f redhat/rpmmacros /tmp/asterisk/ ; \
|
||||
sed "s/Version:/Version: $(RPMVERSION)/g" redhat/asterisk.spec > /tmp/asterisk/asterisk.spec ; \
|
||||
rpm --rcfile /usr/lib/rpm/rpmrc:/tmp/asterisk/rpmrc -bb /tmp/asterisk/asterisk.spec ; \
|
||||
mv /tmp/asterisk/redhat/RPMS/i386/asterisk* ./ ; \
|
||||
rm -rf /tmp/asterisk
|
||||
|
||||
progdocs:
|
||||
(cat contrib/asterisk-ng-doxygen; echo "HAVE_DOT=$(HAVEDOT)"; \
|
||||
echo "PROJECT_NUMBER=$(ASTERISKVERSION)") | doxygen -
|
||||
|
||||
config:
|
||||
@if [ "${OSARCH}" = "linux-gnu" ]; then \
|
||||
if [ -f /etc/redhat-release -o -f /etc/fedora-release ]; then \
|
||||
$(INSTALL) -m 755 contrib/init.d/rc.redhat.asterisk /etc/rc.d/init.d/asterisk; \
|
||||
/sbin/chkconfig --add asterisk; \
|
||||
elif [ -f /etc/debian_version ]; then \
|
||||
$(INSTALL) -m 755 contrib/init.d/rc.debian.asterisk /etc/init.d/asterisk; \
|
||||
/usr/sbin/update-rc.d asterisk start 10 2 3 4 5 . stop 91 2 3 4 5 .; \
|
||||
elif [ -f /etc/gentoo-release ]; then \
|
||||
$(INSTALL) -m 755 contrib/init.d/rc.gentoo.asterisk /etc/init.d/asterisk; \
|
||||
/sbin/rc-update add asterisk default; \
|
||||
elif [ -f /etc/mandrake-release ]; then \
|
||||
$(INSTALL) -m 755 contrib/init.d/rc.mandrake.asterisk /etc/rc.d/init.d/asterisk; \
|
||||
/sbin/chkconfig --add asterisk; \
|
||||
elif [ -f /etc/SuSE-release -o -f /etc/novell-release ]; then \
|
||||
$(INSTALL) -m 755 contrib/init.d/rc.suse.asterisk /etc/init.d/asterisk; \
|
||||
/sbin/chkconfig --add asterisk; \
|
||||
elif [ -f /etc/slackware-version ]; then \
|
||||
echo "Slackware is not currently supported, although an init script does exist for it." \
|
||||
else \
|
||||
echo "We could not install init scripts for your distribution."; \
|
||||
fi \
|
||||
else \
|
||||
echo "We could not install init scripts for your operating system."; \
|
||||
fi
|
||||
|
||||
sounds:
|
||||
$(MAKE) -C sounds all
|
||||
|
||||
# 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
|
||||
|
||||
cleantest:
|
||||
@if ! cmp -s .cleancount .lastclean ; then \
|
||||
$(MAKE) clean;\
|
||||
fi
|
||||
|
||||
$(SUBDIRS_UNINSTALL):
|
||||
@$(MAKE) --no-print-directory -C $(@:-uninstall=) uninstall
|
||||
|
||||
_uninstall: $(SUBDIRS_UNINSTALL)
|
||||
rm -f $(DESTDIR)$(MODULES_DIR)/*
|
||||
rm -f $(DESTDIR)$(ASTSBINDIR)/*asterisk*
|
||||
rm -f $(DESTDIR)$(ASTSBINDIR)/astgenkey
|
||||
rm -f $(DESTDIR)$(ASTSBINDIR)/autosupport
|
||||
rm -rf $(DESTDIR)$(ASTHEADERDIR)
|
||||
rm -rf $(DESTDIR)$(ASTDATADIR)/firmware
|
||||
rm -rf $(DESTDIR)$(ASTMANDIR)/man8
|
||||
$(MAKE) -C sounds uninstall
|
||||
|
||||
uninstall: _uninstall
|
||||
@echo " +--------- Asterisk Uninstall Complete -----+"
|
||||
@echo " + Asterisk binaries, sounds, man pages, +"
|
||||
@echo " + headers, modules, and firmware builds, +"
|
||||
@echo " + have all been uninstalled. +"
|
||||
@echo " + +"
|
||||
@echo " + To remove ALL traces of Asterisk, +"
|
||||
@echo " + including configuration, spool +"
|
||||
@echo " + directories, and logs, run the following +"
|
||||
@echo " + command: +"
|
||||
@echo " + +"
|
||||
@echo " + $(MAKE) uninstall-all +"
|
||||
@echo " +-------------------------------------------+"
|
||||
|
||||
uninstall-all: _uninstall
|
||||
rm -rf $(DESTDIR)$(ASTLIBDIR)
|
||||
rm -rf $(DESTDIR)$(ASTVARLIBDIR)
|
||||
rm -rf $(DESTDIR)$(ASTDATADIR)
|
||||
rm -rf $(DESTDIR)$(ASTSPOOLDIR)
|
||||
rm -rf $(DESTDIR)$(ASTETCDIR)
|
||||
rm -rf $(DESTDIR)$(ASTLOGDIR)
|
||||
|
||||
menuconfig: menuselect
|
||||
|
||||
gmenuconfig: gmenuselect
|
||||
|
||||
menuselect: menuselect/menuselect menuselect-tree
|
||||
-@menuselect/menuselect $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS) menuselect.makeopts && (echo "menuselect changes saved!"; rm -f channels/h323/Makefile.ast main/asterisk) || echo "menuselect changes NOT saved!"
|
||||
|
||||
gmenuselect: menuselect/gmenuselect menuselect-tree
|
||||
-@menuselect/gmenuselect $(GLOBAL_MAKEOPTS) $(USER_MAKEOPTS) menuselect.makeopts && (echo "menuselect changes saved!"; rm -f channels/h323/Makefile.ast main/asterisk) || echo "menuselect changes NOT saved!"
|
||||
|
||||
menuselect/menuselect: makeopts menuselect/menuselect.c menuselect/menuselect_curses.c menuselect/menuselect_stub.c menuselect/menuselect.h menuselect/linkedlists.h makeopts
|
||||
@CC="$(HOST_CC)" LD="" AR="" RANLIB="" CFLAGS="" $(MAKE) -C menuselect CONFIGURE_SILENT="--silent"
|
||||
|
||||
menuselect/gmenuselect: makeopts menuselect/menuselect.c menuselect/menuselect_gtk.c menuselect/menuselect_stub.c menuselect/menuselect.h menuselect/linkedlists.h makeopts
|
||||
@CC="$(HOST_CC)" CXX="$(CXX)" LD="" AR="" RANLIB="" CFLAGS="" $(MAKE) -C menuselect _gmenuselect CONFIGURE_SILENT="--silent"
|
||||
|
||||
menuselect-tree: $(foreach dir,$(filter-out main,$(MOD_SUBDIRS)),$(wildcard $(dir)/*.c) $(wildcard $(dir)/*.cc)) build_tools/cflags.xml sounds/sounds.xml build_tools/embed_modules.xml
|
||||
@echo "Generating input for menuselect ..."
|
||||
@build_tools/prep_moduledeps > $@
|
||||
|
||||
.PHONY: menuselect main sounds clean dist-clean distclean all prereqs cleantest uninstall _uninstall uninstall-all dont-optimize $(SUBDIRS_INSTALL) $(SUBDIRS_CLEAN) $(SUBDIRS_UNINSTALL) $(SUBDIRS) $(MOD_SUBDIRS_EMBED_LDSCRIPT) $(MOD_SUBDIRS_EMBED_LDFLAGS) $(MOD_SUBDIRS_EMBED_LIBS) menuselect.makeopts
|
||||
doxygen asterisk-ng-doxygen
|
||||
|
||||
@@ -1,84 +0,0 @@
|
||||
#
|
||||
# Asterisk -- A telephony toolkit for Linux.
|
||||
#
|
||||
# Makefile rules for subdirectories containing modules
|
||||
#
|
||||
# Copyright (C) 2006, Digium, Inc.
|
||||
#
|
||||
# Kevin P. Fleming <kpfleming@digium.com>
|
||||
#
|
||||
# This program is free software, distributed under the terms of
|
||||
# the GNU General Public License
|
||||
#
|
||||
|
||||
ifneq ($(findstring MALLOC_DEBUG,$(MENUSELECT_CFLAGS)),)
|
||||
ifeq ($(findstring astmm.h,$(ASTCFLAGS)),)
|
||||
ASTCFLAGS+=-include $(ASTTOPDIR)/include/asterisk/astmm.h
|
||||
endif
|
||||
endif
|
||||
|
||||
ifeq ($(findstring LOADABLE_MODULES,$(MENUSELECT_CFLAGS)),)
|
||||
ASTCFLAGS+=${GC_CFLAGS}
|
||||
endif
|
||||
|
||||
ifneq ($(findstring STATIC_BUILD,$(MENUSELECT_CFLAGS)),)
|
||||
STATIC_BUILD=-static
|
||||
endif
|
||||
|
||||
include $(ASTTOPDIR)/Makefile.rules
|
||||
|
||||
comma:=,
|
||||
|
||||
$(addsuffix .o,$(C_MODS)): ASTCFLAGS+=-DAST_MODULE=\"$*\" $(MENUSELECT_OPTS_$*:%=-D%) $(foreach dep,$(MENUSELECT_DEPENDS_$*),$(value $(dep)_INCLUDE))
|
||||
$(addsuffix .oo,$(CC_MODS)): ASTCFLAGS+=-DAST_MODULE=\"$*\" $(MENUSELECT_OPTS_$*:%=-D%) $(foreach dep,$(MENUSELECT_DEPENDS_$*),$(value $(dep)_INCLUDE))
|
||||
|
||||
$(LOADABLE_MODS:%=%.so): ASTCFLAGS+=-fPIC
|
||||
$(LOADABLE_MODS:%=%.so): LIBS+=$(foreach dep,$(MENUSELECT_DEPENDS_$*),$(value $(dep)_LIB))
|
||||
$(LOADABLE_MODS:%=%.so): ASTLDFLAGS+=$(foreach dep,$(MENUSELECT_DEPENDS_$*),$(value $(dep)_LDFLAGS))
|
||||
|
||||
$(addsuffix .so,$(filter $(LOADABLE_MODS),$(C_MODS))): %.so: %.o
|
||||
$(addsuffix .so,$(filter $(LOADABLE_MODS),$(CC_MODS))): %.so: %.oo
|
||||
|
||||
modules.link: $(addsuffix .o,$(filter $(EMBEDDED_MODS),$(C_MODS)))
|
||||
modules.link: $(addsuffix .oo,$(filter $(EMBEDDED_MODS),$(CC_MODS)))
|
||||
|
||||
.PHONY: clean uninstall _all
|
||||
|
||||
ifneq ($(LOADABLE_MODS),)
|
||||
_all: $(LOADABLE_MODS:%=%.so)
|
||||
endif
|
||||
|
||||
ifneq ($(EMBEDDED_MODS),)
|
||||
_all: modules.link
|
||||
__embed_ldscript:
|
||||
@echo "../$(SUBDIR)/modules.link"
|
||||
__embed_ldflags:
|
||||
@echo "$(foreach mod,$(filter $(EMBEDDED_MODS),$(C_MODS)),$(foreach dep,$(MENUSELECT_DEPENDS_$(mod)),$(dep)_LDFLAGS))"
|
||||
@echo "$(foreach mod,$(filter $(EMBEDDED_MODS),$(CC_MODS)),$(foreach dep,$(MENUSELECT_DEPENDS_$(mod)),$(dep)_LDFLAGS))"
|
||||
__embed_libs:
|
||||
@echo "$(foreach mod,$(filter $(EMBEDDED_MODS),$(C_MODS)),$(foreach dep,$(MENUSELECT_DEPENDS_$(mod)),$(dep)_LIB))"
|
||||
@echo "$(foreach mod,$(filter $(EMBEDDED_MODS),$(CC_MODS)),$(foreach dep,$(MENUSELECT_DEPENDS_$(mod)),$(dep)_LIB))"
|
||||
else
|
||||
__embed_ldscript:
|
||||
__embed_ldflags:
|
||||
__embed_libs:
|
||||
endif
|
||||
|
||||
modules.link:
|
||||
@rm -f $@
|
||||
@for file in $(patsubst %,$(SUBDIR)/%,$(filter %.o,$^)); do echo "INPUT (../$${file})" >> $@; done
|
||||
@for file in $(patsubst %,$(SUBDIR)/%,$(filter-out %.o,$^)); do echo "INPUT (../$${file})" >> $@; done
|
||||
|
||||
clean::
|
||||
rm -f *.so *.o *.oo
|
||||
rm -f .*.o.d .*.oo.d
|
||||
rm -f modules.link
|
||||
|
||||
install:: all
|
||||
for x in $(LOADABLE_MODS:%=%.so); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ; done
|
||||
|
||||
uninstall::
|
||||
|
||||
ifneq ($(wildcard .*.d),)
|
||||
include .*.d
|
||||
endif
|
||||
@@ -1,81 +0,0 @@
|
||||
#
|
||||
# Asterisk -- A telephony toolkit for Linux.
|
||||
#
|
||||
# Makefile rules
|
||||
#
|
||||
# Copyright (C) 2006, Digium, Inc.
|
||||
#
|
||||
# Kevin P. Fleming <kpfleming@digium.com>
|
||||
#
|
||||
# This program is free software, distributed under the terms of
|
||||
# the GNU General Public License
|
||||
#
|
||||
|
||||
# Each command is preceded by a short comment on what to do.
|
||||
# Prefixing one or the other with @\# or @ or nothing makes the desired
|
||||
# behaviour. ECHO_PREFIX prefixes the comment, CMD_PREFIX prefixes the command.
|
||||
|
||||
-include $(ASTTOPDIR)/makeopts
|
||||
|
||||
ifeq ($(NOISY_BUILD),)
|
||||
ECHO_PREFIX=@
|
||||
CMD_PREFIX=@
|
||||
else
|
||||
ECHO_PREFIX=@\#
|
||||
CMD_PREFIX=
|
||||
endif
|
||||
|
||||
ifeq ($(findstring DONT_OPTIMIZE,$(MENUSELECT_CFLAGS)),)
|
||||
# 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
|
||||
|
||||
OPTIMIZE?=-O6
|
||||
ASTCFLAGS+=$(OPTIMIZE)
|
||||
endif
|
||||
|
||||
%.o: %.c
|
||||
$(ECHO_PREFIX) echo " [CC] $< -> $@"
|
||||
ifeq ($(AST_DEVMODE),yes)
|
||||
$(CMD_PREFIX) $(CC) -o $@ -c $< $(PTHREAD_CFLAGS) $(ASTCFLAGS) -MMD -MT $@ -MF .$(subst /,_,$@).d -MP
|
||||
else
|
||||
$(CMD_PREFIX) $(CC) -o $@ -c $< $(PTHREAD_CFLAGS) $(ASTCFLAGS)
|
||||
endif
|
||||
|
||||
%.o: %.s
|
||||
$(ECHO_PREFIX) echo " [AS] $< -> $@"
|
||||
ifeq ($(AST_DEVMODE),yes)
|
||||
$(CMD_PREFIX) $(CC) -o $@ -c $< $(PTHREAD_CFLAGS) $(ASTCFLAGS) -MMD -MT $@ -MF .$(subst /,_,$@).d -MP
|
||||
else
|
||||
$(CMD_PREFIX) $(CC) -o $@ -c $< $(PTHREAD_CFLAGS) $(ASTCFLAGS)
|
||||
endif
|
||||
|
||||
%.oo: %.cc
|
||||
$(ECHO_PREFIX) echo " [CXX] $< -> $@"
|
||||
ifeq ($(AST_DEVMODE),yes)
|
||||
$(CMD_PREFIX) $(CXX) -o $@ -c $< $(PTHREAD_CFLAGS) $(filter-out -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations,$(ASTCFLAGS)) -MMD -MT $@ -MF .$(subst /,_,$@).d -MP
|
||||
else
|
||||
$(CMD_PREFIX) $(CXX) -o $@ -c $< $(PTHREAD_CFLAGS) $(filter-out -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations,$(ASTCFLAGS))
|
||||
endif
|
||||
|
||||
%.c: %.y
|
||||
$(ECHO_PREFIX) echo " [BISON] $< -> $@"
|
||||
$(CMD_PREFIX) bison -o $@ -d --name-prefix=ast_yy $<
|
||||
|
||||
%.c: %.fl
|
||||
$(ECHO_PREFIX) echo " [FLEX] $< -> $@"
|
||||
$(CMD_PREFIX) flex -o $@ --full $<
|
||||
|
||||
%.so: %.o
|
||||
$(ECHO_PREFIX) echo " [LD] $^ -> $@"
|
||||
$(CMD_PREFIX) $(CC) $(STATIC_BUILD) -o $@ $(PTHREAD_CFLAGS) $(ASTLDFLAGS) $(SOLINK) $^ $(PTHREAD_LIBS) $(LIBS)
|
||||
|
||||
%.so: %.oo
|
||||
$(ECHO_PREFIX) echo " [LDXX] $^ -> $@"
|
||||
$(CMD_PREFIX) $(CXX) $(STATIC_BUILD) -o $@ $(PTHREAD_CFLAGS) $(ASTLDFLAGS) $(SOLINK) $^ $(PTHREAD_LIBS) $(LIBS)
|
||||
|
||||
%: %.o
|
||||
$(ECHO_PREFIX) echo " [LD] $^ -> $@"
|
||||
$(CMD_PREFIX) $(CXX) $(STATIC_BUILD) -o $@ $(PTHREAD_CFLAGS) $(ASTLDFLAGS) $^ $(PTHREAD_LIBS) $(LIBS)
|
||||
244
README
244
README
@@ -1,155 +1,98 @@
|
||||
The Asterisk(R) Open Source PBX
|
||||
by Mark Spencer <markster@digium.com>
|
||||
and the Asterisk.org developer community
|
||||
|
||||
Copyright (C) 2001-2006 Digium, Inc.
|
||||
and other copyright holders.
|
||||
The Asterisk Open Source PBX
|
||||
by Mark Spencer <markster@linux-support.net>
|
||||
Copyright (C) 2001, Linux Support Services, Inc.
|
||||
================================================================
|
||||
|
||||
* SECURITY
|
||||
It is imperative that you read and fully understand the contents of
|
||||
the security information file (doc/security.txt) before you attempt
|
||||
to configure and run an Asterisk server.
|
||||
the SECURITY file before you attempt to configure 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:
|
||||
* LICENSING
|
||||
Asterisk is distributed under GNU General Public License. The GPL also
|
||||
must apply to all loadable modules as well, except as defined below.
|
||||
|
||||
http://www.voip-info.org/wiki-Asterisk
|
||||
Linux Support Services, Inc. retains copyright to all of the core
|
||||
Asterisk system, and therefore can grant, at its sole discression, the
|
||||
ability for companies, individuals, or organizations to create proprietary
|
||||
or Open Source (but non-GPL'd) modules which may be dynamically linked at
|
||||
runtime with the portions of Asterisk which fall under our copyright
|
||||
umbrella, or are distributed under more flexible licenses than GPL. At
|
||||
this time (5/21/2001) the only component of Asterisk which is covered
|
||||
under GPL and not under our Copyright is the Xing MP3 decoder.
|
||||
|
||||
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.
|
||||
If you wish to use our code in other GPL programs, don't worry -- there
|
||||
is no requirement that you provide the same exemption in your GPL'd
|
||||
products (although if you've written a module for Asterisk we would
|
||||
strongly encourage you to make the same excemption that we do).
|
||||
|
||||
* SUPPORTED OPERATING SYSTEMS
|
||||
If you have any questions, whatsoever, regarding our licensing policy,
|
||||
please contact us.
|
||||
|
||||
* 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.
|
||||
|
||||
== 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 got supported hardware. To use Asterisk right now,
|
||||
you will need one of the following:
|
||||
|
||||
Supported telephony hardware includes:
|
||||
|
||||
* All Wildcard (tm) products from Digium (www.digium.com)
|
||||
* All Wildcard (tm) products from LSS (www.linux-support.net)
|
||||
* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
|
||||
* any full duplex sound card supported by ALSA or OSS
|
||||
* any ISDN card supported by mISDN on Linux (BRI)
|
||||
* The Xorcom AstriBank channel bank
|
||||
* VoiceTronix OpenLine products
|
||||
* Full Duplex Sound Card supported by Linux
|
||||
* Adtran Atlas 800 Plus
|
||||
* ISDN4Linux compatible ISDN card
|
||||
* Tormenta Dual T1 card (www.bsdtelephony.com.mx)
|
||||
|
||||
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 and
|
||||
zaphfc.
|
||||
Assuming you have one of these (most likely the third) you're ready to
|
||||
proceed:
|
||||
|
||||
* UPGRADING FROM AN EARLIER VERSION
|
||||
1) Run "make"
|
||||
2) Run "make install"
|
||||
|
||||
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:
|
||||
|
||||
1) Read this README file.
|
||||
|
||||
There are more documents than this one in the doc/ directory.
|
||||
You may also want to check the configuration files that contain
|
||||
examples and reference guides. They are all in the configs/
|
||||
directory.
|
||||
|
||||
2) Run "./configure"
|
||||
|
||||
Execute the configure script to guess values for system-dependent
|
||||
variables used during compilation.
|
||||
|
||||
3) Run "make menuselect" [optional]
|
||||
|
||||
This is needed if you want to select the modules that will be
|
||||
compiled and to check modules dependencies.
|
||||
|
||||
4) Run "make"
|
||||
|
||||
Assuming the build completes successfully:
|
||||
|
||||
5) Run "make install"
|
||||
|
||||
Each time you update or checkout from the repository, 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:
|
||||
|
||||
6) "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
|
||||
You'll see a bunch of verbose messages fly by your screen as Asterisk
|
||||
initializes (that's the "very very verbose" mode). When it's ready, if
|
||||
you specified the "c" then you'll get a command line console, that looks
|
||||
like this:
|
||||
|
||||
*CLI>
|
||||
|
||||
You can type "help" at any time to get help with the system. For help
|
||||
You can type "help" at any time to get help with the system. For help
|
||||
with a specific command, type "help <command>". To start the PBX using
|
||||
your sound card, you can type "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
|
||||
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,
|
||||
@@ -158,12 +101,12 @@ parameters'. Internally the use of '=' and '=>' is exactly the same, so
|
||||
they're used only to help make the configuration file easier to
|
||||
understand, and do not affect how it is actually parsed.
|
||||
|
||||
Entries of the form 'variable=value' set the value of some parameter in
|
||||
asterisk. For example, in zapata.conf, one might specify:
|
||||
Entries of the form 'variable=value' set the value of some parameter in
|
||||
asterisk. For example, in tormenta.conf, one might specify:
|
||||
|
||||
switchtype=national
|
||||
|
||||
in order to indicate to Asterisk that the switch they are connecting to is
|
||||
In order to indicate to Asterisk that the switch they are connecting to is
|
||||
of the type "national". In general, the parameter will apply to
|
||||
instantiations which occur below its specification. For example, if the
|
||||
configuration file read:
|
||||
@@ -174,89 +117,18 @@ configuration file read:
|
||||
switchtype = dms100
|
||||
channel => 25-47
|
||||
|
||||
the "national" switchtype would be applied to channels one through
|
||||
Then, the "national" switchtype would be applied to channels one through
|
||||
four and channels 10 through 12, whereas the "dms100" switchtype would
|
||||
apply to channels 25 through 47.
|
||||
|
||||
The "object => parameters" instantiates an object with the given
|
||||
The "object => parameters" instantiates an object with the given
|
||||
parameters. For example, the line "channel => 25-47" creates objects for
|
||||
the channels 25 through 47 of the card, obtaining the settings
|
||||
the channels 25 through 47 of the tormenta card, obtaining the settings
|
||||
from the variables specified above.
|
||||
|
||||
* SPECIAL NOTE ON TIME
|
||||
|
||||
Those using SIP phones should be aware that Asterisk is sensitive to
|
||||
large jumps in time. Manually changing the system time using date(1)
|
||||
(or other similar commands) may cause SIP registrations and other
|
||||
internal processes to fail. If your system cannot keep accurate time
|
||||
by itself use NTP (http://www.ntp.org/) to keep the system clock
|
||||
synchronized to "real time". NTP is designed to keep the system clock
|
||||
synchronized by speeding up or slowing down the system clock until it
|
||||
is synchronized to "real time" rather than by jumping the time and
|
||||
causing discontinuities. Most Linux distributions include precompiled
|
||||
versions of NTP. Beware of some time synchronization methods that get
|
||||
the correct real time periodically and then manually set the system
|
||||
clock.
|
||||
|
||||
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
|
||||
|
||||
----
|
||||
Asterisk is a trademark belonging to Digium, inc
|
||||
Mark
|
||||
|
||||
14
README.cdr
Normal file
14
README.cdr
Normal file
@@ -0,0 +1,14 @@
|
||||
Asterisk now generates Call Detail Records. See include/asterisk/cdr.h for
|
||||
all the fields which are recorded. By default, records in comma-separated
|
||||
values will be created in /var/log/asterisk/cdr-csv. You can specify
|
||||
account codes and AMA (Automated Machine Accounting) flags on a per-channel
|
||||
(Zaptel et al) or per-user (IAX) basis to help with accounting. Look
|
||||
at the top of cdr/cdr_csv.c to see the format for the records.
|
||||
|
||||
ONE IMPORTANT NOTE: If you are trying to collect records on IAX to IAX calls
|
||||
you need to be aware that by default, IAX will attempt to transfer calls
|
||||
in this situation (if DTMF is not required). When the transfer is completed
|
||||
the call is dumped from the middle machine and thus the call detail records
|
||||
will report a short call time. If you want detailed records you must
|
||||
turn off IAX transfer, but unless your servers are very close together, you
|
||||
will definitely get a latency hit from doing so.
|
||||
@@ -31,7 +31,7 @@ and H.323, some of which are:
|
||||
* High performance, low overhead protocol
|
||||
When running on low-bandwidth connections, or when running
|
||||
large numbers of calls, optimized bandwidth utilization is
|
||||
imperative. IAX uses only 4 bytes of overhead
|
||||
imperitive. IAX uses only 4 bytes of overhead
|
||||
|
||||
* Internationalization support
|
||||
IAX transmits language information, so that remote PBX
|
||||
@@ -133,12 +133,10 @@ The first line of the "general" section is always:
|
||||
|
||||
Following the first line are a number of other possibilities:
|
||||
|
||||
> bindport = <portnum>
|
||||
> port = <portnum>
|
||||
|
||||
This sets the port that IAX will bind to. The default IAX version 1
|
||||
port number is 5036. For IAX version 2, that is now the default in
|
||||
Asterisk, the default port is 4569.
|
||||
It is recommended that this value not be altered in general.
|
||||
This sets the port that IAX will bind to. The default IAX port number is
|
||||
5036. It is recommended that this value not be altered in general.
|
||||
|
||||
> bindaddr = <ipaddr>
|
||||
|
||||
@@ -172,15 +170,12 @@ disallow the LPC10 codec just because it doesn't sound very good.
|
||||
|
||||
These parameters control the operation of the jitter buffer. The
|
||||
jitterbuffer should always be enabled unless you expect all your
|
||||
connections to be over a LAN.
|
||||
* drop count is the maximum number of voice packets to allow to drop
|
||||
(out of 100). Useful values are 3-10.
|
||||
* maxjitterbuffer is the maximum amount of jitter buffer to permit to be
|
||||
used.
|
||||
* maxexcessbuffer is the maximum amount of excess jitter buffer
|
||||
that is permitted before the jitter buffer is slowly shrunk to eliminate
|
||||
latency.
|
||||
* minexcessbuffer is the minimum amount of excess jitter buffer
|
||||
connections to be over a LAN. The drop count is the maximum number of
|
||||
voice packets to allow to drop (out of 100). Useful values are 3-10. The
|
||||
maxjitterbuffer is the maximum amount of jitter buffer to permit to be
|
||||
used. The "maxexcessbuffer" is the maximum amount of excess jitter buffer
|
||||
that is permitted before the jitter buffer is slowly shrunk to eliminate
|
||||
latency.
|
||||
|
||||
> accountcode = <code>
|
||||
> amaflags = [default|omit|billing|documentation]
|
||||
@@ -203,7 +198,7 @@ these packets, improving voice quality.
|
||||
|
||||
> register => <name>[:<secret>]@<host>[:port]
|
||||
|
||||
Any number of registry entries may be instantiated in the general
|
||||
Any number of registery entries may be instantiated in the general
|
||||
section. Registration allows Asterisk to notify a remote Asterisk server
|
||||
(with a fixed address) what our current address is. In order for
|
||||
registration to work, the remote Asterisk server will need to have a
|
||||
@@ -213,20 +208,13 @@ The name is a required field, and is the remote peer name that we wish to
|
||||
identify ourselves as. A secret may be provided as well. The secret is
|
||||
generally a shared password between the local server and the remote
|
||||
server. However, if the secret is in square brackets ([]'s) then it is
|
||||
interpreted as the name of a RSA key to use. In that case, the local Asterisk
|
||||
interpreted as the name of a key to use. In that case, the local Asterisk
|
||||
server must have the *private* key (/var/lib/asterisk/keys/<name>.key) and
|
||||
the remote server will have to have the corresponding public key.
|
||||
|
||||
The "host" is a required field and is the hostname or IP address of the
|
||||
remote Asterisk server. The port specification is optional and is by
|
||||
default 4569 for iax2 if not specified.
|
||||
|
||||
> notransfer = yes | no
|
||||
|
||||
If an IAX phone calls another IAX phone by using a Asterisk server,
|
||||
Asterisk will transfer the call to go peer to peer. If you do not
|
||||
want this, turn on notransfer with a "yes". This is also settable
|
||||
for peers and users.
|
||||
default 5036 if not specified.
|
||||
|
||||
-------------
|
||||
|
||||
@@ -244,7 +232,7 @@ should be an alphanumeric string.
|
||||
> type = [user|peer|friend]
|
||||
|
||||
This line tells Asterisk how to interpret this entity. Users are things
|
||||
that connect to us, while peers are phones we connect to, and a friend is
|
||||
that connect to us, while peers are people we connect to, and a friend is
|
||||
shorthand for creating a user and a peer with identical information
|
||||
|
||||
----------------
|
||||
@@ -256,7 +244,7 @@ One or more context lines may be specified in a user, thus giving the user
|
||||
access to place calls in the given contexts. Contexts are used by
|
||||
Asterisk to divide dialing plans into logical units each with the ability
|
||||
to have numbers interpreted differently, have their own security model,
|
||||
auxiliary switch handling, and include other contexts. Most users are
|
||||
auxilliary switch handling, and include other contexts. Most users are
|
||||
given access to the default context. Trusted users could be given access
|
||||
to the local context for example.
|
||||
|
||||
@@ -274,10 +262,10 @@ the final result being the decision. For example:
|
||||
would deny anyone in 192.168.0.0 with a netmask of 24 bits (class C),
|
||||
whereas:
|
||||
|
||||
> deny = 192.168.0.0/24
|
||||
> permit = 0.0.0.0/0
|
||||
> deny = 192.168.0.0/255.255.255.0
|
||||
> permit = 0.0.0.0/0.0.0.0
|
||||
|
||||
would not deny anyone since the final rule would permit anyone, thus
|
||||
would not deny anyone since the final rule would permit anyone, thsu
|
||||
overriding the denial.
|
||||
|
||||
If no permit/deny rules are listed, it is assumed that someone may connect
|
||||
@@ -293,9 +281,9 @@ perspective of your server.
|
||||
|
||||
You may select which authentication methods are permitted to be used by
|
||||
the user to authenticate to us. Multiple methods may be specified,
|
||||
separated by commas. If md5 or plaintext authentication is selected, a
|
||||
secret must be provided. If RSA authentication is specified, then one or
|
||||
more key names must be specified with "inkeys"
|
||||
separated by commas. If md5 or plaintext authentication is selected, a
|
||||
secret must be provided. If RSA authentication is specified, then one or
|
||||
more key names must be specifed with "inkeys"
|
||||
|
||||
If no secret is specified and no authentication method is specified, then
|
||||
no authentication will be required.
|
||||
@@ -335,35 +323,4 @@ If the host uses dynamic registration, Asterisk may still be given a
|
||||
default IP address to use when dynamic registration has not been performed
|
||||
or has timed out.
|
||||
|
||||
> peercontext = <context>
|
||||
|
||||
Specifies the context name to be passed to the peer for it to use when routing
|
||||
the call through its dial plan. This entry will be used only if a context
|
||||
is not included in the IAX2 channel name passed to the Dial command.
|
||||
|
||||
> qualify = [yes | no | <value>]
|
||||
|
||||
Qualify turns on checking of availability of the remote peer. If the
|
||||
peer becomes unavailable, no calls are placed to the peer until
|
||||
it is reachable again. This is also helpful in certain NAT situations.
|
||||
|
||||
> jitterbuffer = [yes | no]
|
||||
|
||||
Turns on or off the jitterbuffer for this peer
|
||||
|
||||
> mailbox = <mailbox>[@mailboxcontext]
|
||||
|
||||
Specifies a mailbox to check for voicemail notification.
|
||||
|
||||
> permit = <ipaddr>/<netmask>
|
||||
> deny = <ipaddr>/<netmask>
|
||||
|
||||
Permit and deny rules may be applied to users, allowing them to connect
|
||||
from certain IP addresses and not others. The permit and deny rules are
|
||||
interpreted in sequence and all are evaluated on a given IP address, with
|
||||
the final result being the decision. See the user section above
|
||||
for examples.
|
||||
|
||||
----------------------------------------------------------------------
|
||||
For more examples of a configuration, please see the iax.conf.sample in
|
||||
your the /configs directory of you source code distribution
|
||||
41
SECURITY
Normal file
41
SECURITY
Normal file
@@ -0,0 +1,41 @@
|
||||
==== Security Notes with Asterisk ====
|
||||
|
||||
PLEASE READ THE FOLLOWING IMPORTANT SECURITY RELATED INFORMATION.
|
||||
IMPROPER CONFIGURATION OF ASTERISK COULD ALLOW UNAUTHORIZED USE OF YOUR
|
||||
FACILITIES, POTENTIALLY INCURRING SUBSTANTIAL CHARGES.
|
||||
|
||||
First and foremost remember this:
|
||||
|
||||
USE THE EXTENSION CONTEXTS TO ISOLATE OUTGOING OR TOLL SERVICES FROM ANY
|
||||
INCOMING CONNECTIONS.
|
||||
|
||||
You should consider that if any channel, incoming line, etc can enter an
|
||||
extension context that it has the capability of accessing any extension
|
||||
within that context.
|
||||
|
||||
Therefore, you should NOT allow access to outgoing or toll services in
|
||||
contexts that are accessible (especially without a password) from incoming
|
||||
channels, be they IAX channels, FX or other trunks, or even untrusted
|
||||
stations within you network. In particular, never ever put outgoing toll
|
||||
services in the "default" context. To make things easier, you can include
|
||||
the "default" context within other private contexts by using:
|
||||
|
||||
include => default
|
||||
|
||||
in the appropriate section. A well designed PBX might look like this:
|
||||
|
||||
[longdistance]
|
||||
exten => _91NXXNXXXXXX,1,Dial,Tor/g2/BYEXTENSION
|
||||
include => local
|
||||
|
||||
[local]
|
||||
exten => _9NXXNXXX,1,Dial,Tor/g2/BYEXTENSION
|
||||
include => default
|
||||
|
||||
[default]
|
||||
exten => 6123,Dial,Tor/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.
|
||||
461
UPGRADE.txt
461
UPGRADE.txt
@@ -1,461 +0,0 @@
|
||||
Information for Upgrading From Previous Asterisk Releases
|
||||
=========================================================
|
||||
|
||||
Build Process (configure script):
|
||||
|
||||
Asterisk now uses an autoconf-generated configuration script to learn how it
|
||||
should build itself for your system. As it is a standard script, running:
|
||||
|
||||
$ ./configure --help
|
||||
|
||||
will show you all the options available. This script can be used to tell the
|
||||
build process what libraries you have on your system (if it cannot find them
|
||||
automatically), which libraries you wish to have ignored even though they may
|
||||
be present, etc.
|
||||
|
||||
You must run the configure script before Asterisk will build, although it will
|
||||
attempt to automatically run it for you with no options specified; for most
|
||||
users, that will result in a similar build to what they would have had before
|
||||
the configure script was added to the build process (except for having to run
|
||||
'make' again after the configure script is run). Note that the configure script
|
||||
does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
|
||||
when your system configuration changes or you wish to build Asterisk with
|
||||
different options.
|
||||
|
||||
Build Process (module selection):
|
||||
|
||||
The Asterisk source tree now includes a basic module selection and build option
|
||||
selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
|
||||
In this tool, you can disable building of modules that you don't care about,
|
||||
turn on/off global options for the build and see which modules will not
|
||||
(and cannot) be built because your system does not have the required external
|
||||
dependencies installed.
|
||||
|
||||
The resulting file from menuselect is called 'menuselect.makeopts'. Note that
|
||||
the resulting menuselect.makeopts file generally contains which modules *not*
|
||||
to build. The modules listed in this file indicate which modules have unmet
|
||||
dependencies, a present conflict, or have been disabled by the user in the
|
||||
menuselect interface. Compiler Flags can also be set in the menuselect
|
||||
interface. In this case, the resulting file contains which CFLAGS are in use,
|
||||
not which ones are not in use.
|
||||
|
||||
If you would like to save your choices and have them applied against all
|
||||
builds, the file can be copied to '~/.asterisk.makeopts' or
|
||||
'/etc/asterisk.makeopts'.
|
||||
|
||||
Build Process (Makefile targets):
|
||||
|
||||
The 'valgrind' and 'dont-optimize' targets have been removed; their functionality
|
||||
is available by enabling the DONT_OPTIMIZE setting in the 'Compiler Flags' menu
|
||||
in the menuselect tool.
|
||||
|
||||
It is now possible to run most make targets against a single subdirectory; from
|
||||
the top level directory, for example, 'make channels' will run 'make all' in the
|
||||
'channels' subdirectory. This also is true for 'clean', 'distclean' and 'depend'.
|
||||
|
||||
Sound (prompt) and Music On Hold files:
|
||||
|
||||
Beginning with Asterisk 1.4, the sound files and music on hold files supplied for
|
||||
use with Asterisk have been replaced with new versions produced from high quality
|
||||
master recordings, and are available in three languages (English, French and
|
||||
Spanish) and in five formats (WAV (uncompressed), mu-Law, a-Law, GSM and G.729).
|
||||
In addition, the music on hold files provided by opsound.org Music are now available
|
||||
in the same five formats, but no longer available in MP3 format.
|
||||
|
||||
The Asterisk 1.4 tarball packages will only include English prompts in GSM format,
|
||||
(as were supplied with previous releases) and the opsound.org MOH files in WAV format.
|
||||
All of the other variations can be installed by running 'make menuselect' and
|
||||
selecting the packages you wish to install; when you run 'make install', those
|
||||
packages will be downloaded and installed along with the standard files included
|
||||
in the tarball.
|
||||
|
||||
If for some reason you expect to not have Internet access at the time you will be
|
||||
running 'make install', you can make your package selections using menuselect and
|
||||
then run 'make sounds' to download (only) the sound packages; this will leave the
|
||||
sound packages in the 'sounds' subdirectory to be used later during installation.
|
||||
|
||||
WARNING: Asterisk 1.4 supports a new layout for sound files in multiple languages;
|
||||
instead of the alternate-language files being stored in subdirectories underneath
|
||||
the existing files (for French, that would be digits/fr, letters/fr, phonetic/fr,
|
||||
etc.) the new layout creates one directory under /var/lib/asterisk/sounds for the
|
||||
language itself, then places all the sound files for that language under that
|
||||
directory and its subdirectories. This is the layout that will be created if you
|
||||
select non-English languages to be installed via menuselect, HOWEVER Asterisk does
|
||||
not default to this layout and will not find the files in the places it expects them
|
||||
to be. If you wish to use this layout, make sure you put 'languageprefix=yes' in your
|
||||
/etc/asterisk/asterisk.conf file, so that Asterisk will know how the files were
|
||||
installed.
|
||||
|
||||
PBX Core:
|
||||
|
||||
* The (very old and undocumented) ability to use BYEXTENSION for dialing
|
||||
instead of ${EXTEN} has been removed.
|
||||
|
||||
* Builtin (res_features) transfer functionality attempts to use the context
|
||||
defined in TRANSFER_CONTEXT variable of the transferer channel first. If
|
||||
not set, it uses the transferee variable. If not set in any channel, it will
|
||||
attempt to use the last non macro context. If not possible, it will default
|
||||
to the current context.
|
||||
|
||||
* The autofallthrough setting introduced in Asterisk 1.2 now defaults to 'yes';
|
||||
if your dialplan relies on the ability to 'run off the end' of an extension
|
||||
and wait for a new extension without using WaitExten() to accomplish that,
|
||||
you will need set autofallthrough to 'no' in your extensions.conf file.
|
||||
|
||||
Command Line Interface:
|
||||
|
||||
* 'show channels concise', designed to be used by applications that will parse
|
||||
its output, previously used ':' characters to separate fields. However, some
|
||||
of those fields can easily contain that character, making the output not
|
||||
parseable. The delimiter has been changed to '!'.
|
||||
|
||||
Applications:
|
||||
|
||||
* In previous Asterisk releases, many applications would jump to priority n+101
|
||||
to indicate some kind of status or error condition. This functionality was
|
||||
marked deprecated in Asterisk 1.2. An option to disable it was provided with
|
||||
the default value set to 'on'. The default value for the global priority
|
||||
jumping option is now 'off'.
|
||||
|
||||
* The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
|
||||
AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
|
||||
and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
|
||||
been removed in this version. You should use the equivalent dialplan
|
||||
function in places where you have previously used one of these applications.
|
||||
|
||||
* The application SetGlobalVar has been deprecated. You should replace uses
|
||||
of this application with the following combination of Set and GLOBAL():
|
||||
Set(GLOBAL(name)=value). You may also access global variables exclusively by
|
||||
using the GLOBAL() dialplan function, instead of relying on variable
|
||||
interpolation falling back to globals when no channel variable is set.
|
||||
|
||||
* The application SetVar has been renamed to Set. The syntax SetVar was marked
|
||||
deprecated in version 1.2 and is no longer recognized in this version.
|
||||
|
||||
* app_read has been updated to use the newer options codes, using "skip" or
|
||||
"noanswer" will not work. Use s or n. Also there is a new feature i, for
|
||||
using indication tones, so typing in skip would give you unexpected results.
|
||||
|
||||
* OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.
|
||||
|
||||
* The CONNECT event in the queue_log from app_queue now has a second field
|
||||
in addition to the holdtime field. It contains the unique ID of the
|
||||
queue member channel that is taking the call. This is useful when trying
|
||||
to link recording filenames back to a particular call from the queue.
|
||||
|
||||
* The old/current behavior of app_queue has a serial type behavior
|
||||
in that the queue will make all waiting callers wait in the queue
|
||||
even if there is more than one available member ready to take
|
||||
calls until the head caller is connected with the member they
|
||||
were trying to get to. The next waiting caller in line then
|
||||
becomes the head caller, and they are then connected with the
|
||||
next available member and all available members and waiting callers
|
||||
waits while this happens. This cycle continues until there are
|
||||
no more available members or waiting callers, whichever comes first.
|
||||
The new behavior, enabled by setting autofill=yes in queues.conf
|
||||
either at the [general] level to default for all queues or
|
||||
to set on a per-queue level, makes sure that when the waiting
|
||||
callers are connecting with available members in a parallel fashion
|
||||
until there are no more available members or no more waiting callers,
|
||||
whichever comes first. This is probably more along the lines of how
|
||||
one would expect a queue should work and in most cases, you will want
|
||||
to enable this new behavior. If you do not specify or comment out this
|
||||
option, it will default to "no" to keep backward compatability with the old
|
||||
behavior.
|
||||
|
||||
* Queues depend on the channel driver reporting the proper state
|
||||
for each member of the queue. To get proper signalling on
|
||||
queue members that use the SIP channel driver, you need to
|
||||
enable a call limit (could be set to a high value so it
|
||||
is not put into action) and also make sure that both inbound
|
||||
and outbound calls are accounted for.
|
||||
|
||||
Example:
|
||||
|
||||
[general]
|
||||
limitonpeer = yes
|
||||
|
||||
[peername]
|
||||
type=friend
|
||||
call-limit=10
|
||||
|
||||
|
||||
* The app_queue application now has the ability to use MixMonitor to
|
||||
record conversations queue members are having with queue callers. Please
|
||||
see configs/queues.conf.sample for more information on this option.
|
||||
|
||||
* The app_queue application strategy called 'roundrobin' has been deprecated
|
||||
for this release. Users are encouraged to use 'rrmemory' instead, since it
|
||||
provides more 'true' round-robin call delivery. For the Asterisk 1.6 release,
|
||||
'rrmemory' will be renamed 'roundrobin'.
|
||||
|
||||
* app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
|
||||
the 'm' option now provides the functionality of "initially muted".
|
||||
In practice, most existing dialplans using the 'm' flag should not notice
|
||||
any difference, unless the keypad menu is enabled, allowing the user
|
||||
to unmute themsleves.
|
||||
|
||||
* ast_play_and_record would attempt to cancel the recording if a DTMF
|
||||
'0' was received. This behavior was not documented in most of the
|
||||
applications that used ast_play_and_record and the return codes from
|
||||
ast_play_and_record weren't checked for properly.
|
||||
ast_play_and_record has been changed so that '0' no longer cancels a
|
||||
recording. If you want to allow DTMF digits to cancel an
|
||||
in-progress recording use ast_play_and_record_full which allows you
|
||||
to specify which DTMF digits can be used to accept a recording and
|
||||
which digits can be used to cancel a recording.
|
||||
|
||||
* ast_app_messagecount has been renamed to ast_app_inboxcount. There is now a
|
||||
new ast_app_messagecount function which takes a single context/mailbox/folder
|
||||
mailbox specification and returns the message count for that folder only.
|
||||
This addresses the deficiency of not being able to count the number of
|
||||
messages in folders other than INBOX and Old.
|
||||
|
||||
* The exit behavior of the AGI applications has changed. Previously, when
|
||||
a connection to an AGI server failed, the application would cause the channel
|
||||
to immediately stop dialplan execution and hangup. Now, the only time that
|
||||
the AGI applications will cause the channel to stop dialplan execution is
|
||||
when the channel itself requests hangup. The AGI applications now set an
|
||||
AGISTATUS variable which will allow you to find out whether running the AGI
|
||||
was successful or not.
|
||||
|
||||
Previously, there was no way to handle the case where Asterisk was unable to
|
||||
locally execute an AGI script for some reason. In this case, dialplan
|
||||
execution will continue as it did before, but the AGISTATUS variable will be
|
||||
set to "FAILURE".
|
||||
|
||||
A locally executed AGI script can now exit with a non-zero exit code and this
|
||||
failure will be detected by Asterisk. If an AGI script exits with a non-zero
|
||||
exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
|
||||
"SUCCESS".
|
||||
|
||||
* app_voicemail: The ODBC_STORAGE capability now requires the extended table format
|
||||
previously used only by EXTENDED_ODBC_STORAGE. This means that you will need to update
|
||||
your table format using the schema provided in doc/odbcstorage.txt
|
||||
|
||||
* app_waitforsilence: Fixes have been made to this application which changes the
|
||||
default behavior with how quickly it returns. You can maintain "old-style" behavior
|
||||
with the addition/use of a third "timeout" parameter.
|
||||
Please consult the application documentation and make changes to your dialplan
|
||||
if appropriate.
|
||||
|
||||
Manager:
|
||||
|
||||
* After executing the 'status' manager action, the "Status" manager events
|
||||
included the header "CallerID:" which was actually only the CallerID number,
|
||||
and not the full CallerID string. This header has been renamed to
|
||||
"CallerIDNum". For compatibility purposes, the CallerID parameter will remain
|
||||
until after the release of 1.4, when it will be removed. Please use the time
|
||||
during the 1.4 release to make this transition.
|
||||
|
||||
* The AgentConnect event now has an additional field called "BridgedChannel"
|
||||
which contains the unique ID of the queue member channel that is taking the
|
||||
call. This is useful when trying to link recording filenames back to
|
||||
a particular call from the queue.
|
||||
|
||||
* app_userevent has been modified to always send Event: UserEvent with the
|
||||
additional header UserEvent: <userspec>. Also, the Channel and UniqueID
|
||||
headers are not automatically sent, unless you specify them as separate
|
||||
arguments. Please see the application help for the new syntax.
|
||||
|
||||
* app_meetme: Mute and Unmute events are now reported via the Manager API.
|
||||
Native Manager API commands MeetMeMute and MeetMeUnmute are provided, which
|
||||
are easier to use than "Action Command:". The MeetMeStopTalking event has
|
||||
also been deprecated in favor of the already existing MeetmeTalking event
|
||||
with a "Status" of "on" or "off" added.
|
||||
|
||||
* OriginateFailure and OriginateSuccess events were replaced by event
|
||||
OriginateResponse with a header named "Response" to indicate success or
|
||||
failure
|
||||
|
||||
Variables:
|
||||
|
||||
* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
|
||||
${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
|
||||
and ${LANGUAGE} have all been deprecated in favor of their related dialplan
|
||||
functions. You are encouraged to move towards the associated dialplan
|
||||
function, as these variables will be removed in a future release.
|
||||
|
||||
* The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now
|
||||
adjustable from cdr.conf, instead of recompiling.
|
||||
|
||||
* OSP applications exports several new variables, ${OSPINHANDLE},
|
||||
${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
|
||||
${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}
|
||||
|
||||
* Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
|
||||
created channel. This variables holds the channel name of the transferer.
|
||||
|
||||
* The dial plan variable PRI_CAUSE will be removed from future versions
|
||||
of Asterisk.
|
||||
It is replaced by adding a cause value to the hangup() application.
|
||||
|
||||
Functions:
|
||||
|
||||
* The function ${CHECK_MD5()} has been deprecated in favor of using an
|
||||
expression: $[${MD5(<string>)} = ${saved_md5}].
|
||||
|
||||
* The 'builtin' functions that used to be combined in pbx_functions.so are
|
||||
now built as separate modules. If you are not using 'autoload=yes' in your
|
||||
modules.conf file then you will need to explicitly load the modules that
|
||||
contain the functions you want to use.
|
||||
|
||||
* The ENUMLOOKUP() function with the 'c' option (for counting the number of
|
||||
records), but the lookup fails to match any records, the returned value will
|
||||
now be "0" instead of blank.
|
||||
|
||||
* The REALTIME() function is now available in version 1.4 and app_realtime has
|
||||
been deprecated in favor of the new function. app_realtime will be removed
|
||||
completely with the version 1.6 release so please take the time between
|
||||
releases to make any necessary changes
|
||||
|
||||
* The QUEUEAGENTCOUNT() function has been deprecated in favor of
|
||||
QUEUE_MEMBER_COUNT().
|
||||
|
||||
The IAX2 channel:
|
||||
|
||||
* The "mailboxdetail" option has been deprecated. Previously, if this option
|
||||
was not enabled, the 2 byte MSGCOUNT information element would be set to all
|
||||
1's to indicate there there is some number of messages waiting. With this
|
||||
option enabled, the number of new messages were placed in one byte and the
|
||||
number of old messages are placed in the other. This is now the default
|
||||
(and the only) behavior.
|
||||
|
||||
The SIP channel:
|
||||
|
||||
* The "incominglimit" setting is replaced by the "call-limit" setting in
|
||||
sip.conf.
|
||||
|
||||
* OSP support code is removed from SIP channel to OSP applications. ospauth
|
||||
option in sip.conf is removed to osp.conf as authpolicy. allowguest option
|
||||
in sip.conf cannot be set as osp anymore.
|
||||
|
||||
* The Asterisk RTP stack has been changed in regards to RFC2833 reception
|
||||
and transmission. Packets will now be sent with proper duration instead of all
|
||||
at once. If you are receiving calls from a pre-1.4 Asterisk installation you
|
||||
will want to turn on the rfc2833compensate option. Without this option your
|
||||
DTMF reception may act poorly.
|
||||
|
||||
* The $SIPUSERAGENT dialplan variable is deprecated and will be removed
|
||||
in coming versions of Asterisk. Please use the dialplan function
|
||||
SIPCHANINFO(useragent) instead.
|
||||
|
||||
* The ALERT_INFO dialplan variable is deprecated and will be removed
|
||||
in coming versions of Asterisk. Please use the dialplan application
|
||||
sipaddheader() to add the "Alert-Info" header to the outbound invite.
|
||||
|
||||
* The "canreinvite" option has changed. canreinvite=yes used to disable
|
||||
re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat
|
||||
to disable re-invites when NAT=yes. This is propably what you want.
|
||||
The settings are now: "yes", "no", "nonat", "update". Please consult
|
||||
sip.conf.sample for detailed information.
|
||||
|
||||
The Zap channel:
|
||||
|
||||
* Support for MFC/R2 has been removed, as it has not been functional for some
|
||||
time and it has no maintainer.
|
||||
|
||||
The Agent channel:
|
||||
|
||||
* Callback mode (AgentCallbackLogin) is now deprecated, since the entire function
|
||||
it provided can be done using dialplan logic, without requiring additional
|
||||
channel and module locks (which frequently caused deadlocks). An example of
|
||||
how to do this using AEL dialplan is in doc/queues-with-callback-members.txt.
|
||||
|
||||
The G726-32 codec:
|
||||
|
||||
* It has been determined that previous versions of Asterisk used the wrong codeword
|
||||
packing order for G726-32 data. This version supports both available packing orders,
|
||||
and can transcode between them. It also now selects the proper order when
|
||||
negotiating with a SIP peer based on the codec name supplied in the SDP. However,
|
||||
there are existing devices that improperly request one order and then use another;
|
||||
Sipura and Grandstream ATAs are known to do this, and there may be others. To
|
||||
be able to continue to use these devices with this version of Asterisk and the
|
||||
G726-32 codec, a configuration parameter called 'g726nonstandard' has been added
|
||||
to sip.conf, so that Asterisk can use the packing order expected by the device (even
|
||||
though it requested a different order). In addition, the internal format number for
|
||||
G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The
|
||||
result of this is that this version of Asterisk will be able to interoperate over
|
||||
IAX2 with older versions of Asterisk, as long as this version is told to allow
|
||||
'g726aal2' instead of 'g726' as the codec for the call.
|
||||
|
||||
Installation:
|
||||
|
||||
* On BSD systems, the installation directories have changed to more "FreeBSDish"
|
||||
directories. On startup, Asterisk will look for the main configuration in
|
||||
/usr/local/etc/asterisk/asterisk.conf
|
||||
If you have an old installation, you might want to remove the binaries and
|
||||
move the configuration files to the new locations. The following directories
|
||||
are now default:
|
||||
ASTLIBDIR /usr/local/lib/asterisk
|
||||
ASTVARLIBDIR /usr/local/share/asterisk
|
||||
ASTETCDIR /usr/local/etc/asterisk
|
||||
ASTBINDIR /usr/local/bin/asterisk
|
||||
ASTSBINDIR /usr/local/sbin/asterisk
|
||||
|
||||
Music on Hold:
|
||||
|
||||
* The music on hold handling has been changed in some significant ways in hopes
|
||||
to make it work in a way that is much less confusing to users. Behavior will
|
||||
not change if the same configuration is used from older versions of Asterisk.
|
||||
However, there are some new configuration options that will make things work
|
||||
in a way that makes more sense.
|
||||
|
||||
Previously, many of the channel drivers had an option called "musicclass" or
|
||||
something similar. This option set what music on hold class this channel
|
||||
would *hear* when put on hold. Some people expected (with good reason) that
|
||||
this option was to configure what music on hold class to play when putting
|
||||
the bridged channel on hold. This option has now been deprecated.
|
||||
|
||||
Two new music on hold related configuration options for channel drivers have
|
||||
been introduced. Some channel drivers support both options, some just one,
|
||||
and some support neither of them. Check the sample configuration files to see
|
||||
which options apply to which channel driver.
|
||||
|
||||
The "mohsuggest" option specifies which music on hold class to suggest to the
|
||||
bridged channel when putting them on hold. The only way that this class can
|
||||
be overridden is if the bridged channel has a specific music class set that
|
||||
was done in the dialplan using Set(CHANNEL(musicclass)=something).
|
||||
|
||||
The "mohinterpret" option is similar to the old "musicclass" option. It
|
||||
specifies which music on hold class this channel would like to listen to when
|
||||
put on hold. This music class is only effective if this channel has no music
|
||||
class set on it from the dialplan and the bridged channel putting this one on
|
||||
hold had no "mohsuggest" setting.
|
||||
|
||||
The IAX2 and Zap channel drivers have an additional feature for the
|
||||
"mohinterpret" option. If this option is set to "passthrough", then these
|
||||
channel drivers will pass through the HOLD message in signalling instead of
|
||||
starting music on hold on the channel. An example for how this would be
|
||||
useful is in an enterprise network of Asterisk servers. When one phone on one
|
||||
server puts a phone on a different server on hold, the remote server will be
|
||||
responsible for playing the hold music to its local phone that was put on
|
||||
hold instead of the far end server across the network playing the music.
|
||||
|
||||
CDR Records:
|
||||
|
||||
* The behavior of the "clid" field of the CDR has always been that it will
|
||||
contain the callerid ANI if it is set, or the callerid number if ANI was not
|
||||
set. When using the "callerid" option for various channel drivers, some
|
||||
would set ANI and some would not. This has been cleared up so that all
|
||||
channel drivers set ANI. If you would like to change the callerid number
|
||||
on the channel from the dialplan and have that change also show up in the
|
||||
CDR, then you *must* set CALLERID(ANI) as well as CALLERID(num).
|
||||
|
||||
API:
|
||||
|
||||
* There are some API functions that were not previously prefixed with the 'ast_'
|
||||
prefix but now are; these include the ADSI, ODBC and AGI interfaces. If you
|
||||
have a module that uses the services provided by res_adsi, res_odbc, or
|
||||
res_agi, you will need to add ast_ prefixes to the functions that you call
|
||||
from those modules.
|
||||
|
||||
Formats:
|
||||
|
||||
* format_wav: The GAIN preprocessor definition has been changed from 2 to 0
|
||||
in Asterisk 1.4. This change was made in response to user complaints of
|
||||
choppiness or the clipping of loud signal peaks. The GAIN preprocessor
|
||||
definition will be retained in Asterisk 1.4, but will be removed in a
|
||||
future release. The use of GAIN for the increasing of voicemail message
|
||||
volume should use the 'volgain' option in voicemail.conf
|
||||
|
||||
939
acinclude.m4
939
acinclude.m4
@@ -1,939 +0,0 @@
|
||||
# AST_GCC_ATTRIBUTE([attribute name])
|
||||
|
||||
AC_DEFUN([AST_GCC_ATTRIBUTE],
|
||||
[
|
||||
AC_MSG_CHECKING(for compiler 'attribute $1' support)
|
||||
AC_COMPILE_IFELSE(
|
||||
AC_LANG_PROGRAM([static int __attribute__(($1)) test(void) {}],
|
||||
[]),
|
||||
AC_MSG_RESULT(yes)
|
||||
AC_DEFINE_UNQUOTED([HAVE_ATTRIBUTE_$1], 1, [Define to 1 if your GCC C compiler supports the '$1' attribute.]),
|
||||
AC_MSG_RESULT(no))
|
||||
])
|
||||
|
||||
# AST_EXT_LIB_SETUP([package symbol name], [package friendly name], [package option name], [additional help text])
|
||||
|
||||
AC_DEFUN([AST_EXT_LIB_SETUP],
|
||||
[
|
||||
$1_DESCRIP="$2"
|
||||
$1_OPTION="$3"
|
||||
AC_ARG_WITH([$3], AC_HELP_STRING([--with-$3=PATH],[use $2 files in PATH $4]),[
|
||||
case ${withval} in
|
||||
n|no)
|
||||
USE_$1=no
|
||||
;;
|
||||
y|ye|yes)
|
||||
$1_MANDATORY="yes"
|
||||
;;
|
||||
*)
|
||||
$1_DIR="${withval}"
|
||||
$1_MANDATORY="yes"
|
||||
;;
|
||||
esac
|
||||
])
|
||||
PBX_$1=0
|
||||
AC_SUBST([$1_LIB])
|
||||
AC_SUBST([$1_INCLUDE])
|
||||
AC_SUBST([$1_DIR])
|
||||
AC_SUBST([PBX_$1])
|
||||
])
|
||||
|
||||
# AST_EXT_LIB_CHECK([package symbol name], [package library name], [function to check], [package header], [additional LIB data])
|
||||
|
||||
AC_DEFUN([AST_EXT_LIB_CHECK],
|
||||
[
|
||||
if test "${USE_$1}" != "no"; then
|
||||
pbxlibdir=""
|
||||
if test "x${$1_DIR}" != "x"; then
|
||||
if test -d ${$1_DIR}/lib; then
|
||||
pbxlibdir="-L${$1_DIR}/lib"
|
||||
else
|
||||
pbxlibdir="-L${$1_DIR}"
|
||||
fi
|
||||
fi
|
||||
AC_CHECK_LIB([$2], [$3], [AST_$1_FOUND=yes], [AST_$1_FOUND=no], ${pbxlibdir} $5)
|
||||
|
||||
if test "${AST_$1_FOUND}" = "yes"; then
|
||||
$1_LIB="-l$2 $5"
|
||||
$1_HEADER_FOUND="1"
|
||||
if test "x${$1_DIR}" != "x"; then
|
||||
$1_LIB="${pbxlibdir} ${$1_LIB}"
|
||||
$1_INCLUDE="-I${$1_DIR}/include"
|
||||
saved_cppflags="${CPPFLAGS}"
|
||||
CPPFLAGS="${CPPFLAGS} -I${$1_DIR}/include"
|
||||
if test "x$4" != "x" ; then
|
||||
AC_CHECK_HEADER([${$1_DIR}/include/$4], [$1_HEADER_FOUND=1], [$1_HEADER_FOUND=0])
|
||||
fi
|
||||
CPPFLAGS="${saved_cppflags}"
|
||||
else
|
||||
if test "x$4" != "x" ; then
|
||||
AC_CHECK_HEADER([$4], [$1_HEADER_FOUND=1], [$1_HEADER_FOUND=0])
|
||||
fi
|
||||
fi
|
||||
if test "x${$1_HEADER_FOUND}" = "x0" ; then
|
||||
if test -n "${$1_MANDATORY}" ;
|
||||
then
|
||||
AC_MSG_NOTICE([***])
|
||||
AC_MSG_NOTICE([*** It appears that you do not have the $2 development package installed.])
|
||||
AC_MSG_NOTICE([*** Please install it to include ${$1_DESCRIP} support, or re-run configure])
|
||||
AC_MSG_NOTICE([*** without explicitly specifying --with-${$1_OPTION}])
|
||||
exit 1
|
||||
fi
|
||||
$1_LIB=""
|
||||
$1_INCLUDE=""
|
||||
PBX_$1=0
|
||||
else
|
||||
PBX_$1=1
|
||||
AC_DEFINE_UNQUOTED([HAVE_$1], 1, [Define to indicate the ${$1_DESCRIP} library])
|
||||
fi
|
||||
elif test -n "${$1_MANDATORY}";
|
||||
then
|
||||
AC_MSG_NOTICE([***])
|
||||
AC_MSG_NOTICE([*** The ${$1_DESCRIP} installation on this system appears to be broken.])
|
||||
AC_MSG_NOTICE([*** Either correct the installation, or run configure])
|
||||
AC_MSG_NOTICE([*** without explicitly specifying --with-${$1_OPTION}])
|
||||
exit 1
|
||||
fi
|
||||
fi
|
||||
])
|
||||
|
||||
|
||||
AC_DEFUN(
|
||||
[AST_CHECK_GNU_MAKE], [AC_CACHE_CHECK(for GNU make, GNU_MAKE,
|
||||
GNU_MAKE='Not Found' ;
|
||||
GNU_MAKE_VERSION_MAJOR=0 ;
|
||||
GNU_MAKE_VERSION_MINOR=0 ;
|
||||
for a in make gmake gnumake ; do
|
||||
if test -z "$a" ; then continue ; fi ;
|
||||
if ( sh -c "$a --version" 2> /dev/null | grep GNU 2>&1 > /dev/null ) ; then
|
||||
GNU_MAKE=$a ;
|
||||
GNU_MAKE_VERSION_MAJOR=`$GNU_MAKE --version | grep "GNU Make" | cut -f3 -d' ' | cut -f1 -d'.'`
|
||||
GNU_MAKE_VERSION_MINOR=`$GNU_MAKE --version | grep "GNU Make" | cut -f2 -d'.' | cut -c1-2`
|
||||
break;
|
||||
fi
|
||||
done ;
|
||||
) ;
|
||||
if test "x$GNU_MAKE" = "xNot Found" ; then
|
||||
AC_MSG_ERROR( *** Please install GNU make. It is required to build Asterisk!)
|
||||
exit 1
|
||||
fi
|
||||
AC_SUBST([GNU_MAKE])
|
||||
])
|
||||
|
||||
|
||||
AC_DEFUN(
|
||||
[AST_CHECK_PWLIB], [
|
||||
PWLIB_INCDIR=
|
||||
PWLIB_LIBDIR=
|
||||
if test "${PWLIBDIR:-unset}" != "unset" ; then
|
||||
AC_CHECK_FILE(${PWLIBDIR}/version.h, HAS_PWLIB=1, )
|
||||
fi
|
||||
if test "${HAS_PWLIB:-unset}" = "unset" ; then
|
||||
if test "${OPENH323DIR:-unset}" != "unset"; then
|
||||
AC_CHECK_FILE(${OPENH323DIR}/../pwlib/version.h, HAS_PWLIB=1, )
|
||||
fi
|
||||
if test "${HAS_PWLIB:-unset}" != "unset" ; then
|
||||
PWLIBDIR="${OPENH323DIR}/../pwlib"
|
||||
else
|
||||
AC_CHECK_FILE(${HOME}/pwlib/include/ptlib.h, HAS_PWLIB=1, )
|
||||
if test "${HAS_PWLIB:-unset}" != "unset" ; then
|
||||
PWLIBDIR="${HOME}/pwlib"
|
||||
else
|
||||
AC_CHECK_FILE(/usr/local/include/ptlib.h, HAS_PWLIB=1, )
|
||||
if test "${HAS_PWLIB:-unset}" != "unset" ; then
|
||||
AC_PATH_PROG(PTLIB_CONFIG, ptlib-config, , /usr/local/bin)
|
||||
if test "${PTLIB_CONFIG:-unset}" = "unset" ; then
|
||||
AC_PATH_PROG(PTLIB_CONFIG, ptlib-config, , /usr/local/share/pwlib/make)
|
||||
fi
|
||||
PWLIB_INCDIR="/usr/local/include"
|
||||
PWLIB_LIBDIR=`${PTLIB_CONFIG} --pwlibdir`
|
||||
if test "${PWLIB_LIBDIR:-unset}" = "unset"; then
|
||||
if test "x$LIB64" != "x"; then
|
||||
PWLIB_LIBDIR="/usr/local/lib64"
|
||||
else
|
||||
PWLIB_LIBDIR="/usr/local/lib"
|
||||
fi
|
||||
fi
|
||||
PWLIB_LIB=`${PTLIB_CONFIG} --ldflags --libs`
|
||||
PWLIB_LIB="-L${PWLIB_LIBDIR} `echo ${PWLIB_LIB}`"
|
||||
else
|
||||
AC_CHECK_FILE(/usr/include/ptlib.h, HAS_PWLIB=1, )
|
||||
if test "${HAS_PWLIB:-unset}" != "unset" ; then
|
||||
AC_PATH_PROG(PTLIB_CONFIG, ptlib-config, , /usr/share/pwlib/make)
|
||||
PWLIB_INCDIR="/usr/include"
|
||||
PWLIB_LIBDIR=`${PTLIB_CONFIG} --pwlibdir`
|
||||
if test "${PWLIB_LIBDIR:-unset}" = "unset"; then
|
||||
if test "x$LIB64" != "x"; then
|
||||
PWLIB_LIBDIR="/usr/lib64"
|
||||
else
|
||||
PWLIB_LIBDIR="/usr/lib"
|
||||
fi
|
||||
fi
|
||||
PWLIB_LIB=`${PTLIB_CONFIG} --ldflags --libs`
|
||||
PWLIB_LIB="-L${PWLIB_LIBDIR} `echo ${PWLIB_LIB}`"
|
||||
fi
|
||||
fi
|
||||
fi
|
||||
fi
|
||||
fi
|
||||
|
||||
#if test "${HAS_PWLIB:-unset}" = "unset" ; then
|
||||
# echo "Cannot find pwlib - please install or set PWLIBDIR and try again"
|
||||
# exit
|
||||
#fi
|
||||
|
||||
if test "${HAS_PWLIB:-unset}" != "unset" ; then
|
||||
if test "${PWLIBDIR:-unset}" = "unset" ; then
|
||||
if test "${PTLIB_CONFIG:-unset}" != "unset" ; then
|
||||
PWLIBDIR=`$PTLIB_CONFIG --prefix`
|
||||
else
|
||||
echo "Cannot find ptlib-config - please install and try again"
|
||||
exit
|
||||
fi
|
||||
fi
|
||||
|
||||
if test "x$PWLIBDIR" = "x/usr" -o "x$PWLIBDIR" = "x/usr/"; then
|
||||
PWLIBDIR="/usr/share/pwlib"
|
||||
PWLIB_INCDIR="/usr/include"
|
||||
if test "x$LIB64" != "x"; then
|
||||
PWLIB_LIBDIR="/usr/lib64"
|
||||
else
|
||||
PWLIB_LIBDIR="/usr/lib"
|
||||
fi
|
||||
fi
|
||||
if test "x$PWLIBDIR" = "x/usr/local" -o "x$PWLIBDIR" = "x/usr/"; then
|
||||
PWLIBDIR="/usr/local/share/pwlib"
|
||||
PWLIB_INCDIR="/usr/local/include"
|
||||
if test "x$LIB64" != "x"; then
|
||||
PWLIB_LIBDIR="/usr/local/lib64"
|
||||
else
|
||||
PWLIB_LIBDIR="/usr/local/lib"
|
||||
fi
|
||||
fi
|
||||
|
||||
if test "${PWLIB_INCDIR:-unset}" = "unset"; then
|
||||
PWLIB_INCDIR="${PWLIBDIR}/include"
|
||||
fi
|
||||
if test "${PWLIB_LIBDIR:-unset}" = "unset"; then
|
||||
PWLIB_LIBDIR="${PWLIBDIR}/lib"
|
||||
fi
|
||||
|
||||
AC_SUBST([PWLIBDIR])
|
||||
AC_SUBST([PWLIB_INCDIR])
|
||||
AC_SUBST([PWLIB_LIBDIR])
|
||||
fi
|
||||
])
|
||||
|
||||
|
||||
AC_DEFUN(
|
||||
[AST_CHECK_OPENH323_PLATFORM], [
|
||||
PWLIB_OSTYPE=
|
||||
case "$host_os" in
|
||||
linux*) PWLIB_OSTYPE=linux ;
|
||||
;;
|
||||
freebsd* ) PWLIB_OSTYPE=FreeBSD ;
|
||||
;;
|
||||
openbsd* ) PWLIB_OSTYPE=OpenBSD ;
|
||||
ENDLDLIBS="-lossaudio" ;
|
||||
;;
|
||||
netbsd* ) PWLIB_OSTYPE=NetBSD ;
|
||||
ENDLDLIBS="-lossaudio" ;
|
||||
;;
|
||||
solaris* | sunos* ) PWLIB_OSTYPE=solaris ;
|
||||
;;
|
||||
darwin* ) PWLIB_OSTYPE=Darwin ;
|
||||
;;
|
||||
beos*) PWLIB_OSTYPE=beos ;
|
||||
STDCCFLAGS="$STDCCFLAGS -D__BEOS__"
|
||||
;;
|
||||
cygwin*) PWLIB_OSTYPE=cygwin ;
|
||||
;;
|
||||
mingw*) PWLIB_OSTYPE=mingw ;
|
||||
STDCCFLAGS="$STDCCFLAGS -mms-bitfields" ;
|
||||
ENDLDLIBS="-lwinmm -lwsock32 -lsnmpapi -lmpr -lcomdlg32 -lgdi32 -lavicap32" ;
|
||||
;;
|
||||
* ) PWLIB_OSTYPE="$host_os" ;
|
||||
AC_MSG_WARN("OS $PWLIB_OSTYPE not recognized - proceed with caution!") ;
|
||||
;;
|
||||
esac
|
||||
|
||||
PWLIB_MACHTYPE=
|
||||
case "$host_cpu" in
|
||||
x86 | i686 | i586 | i486 | i386 ) PWLIB_MACHTYPE=x86
|
||||
;;
|
||||
|
||||
x86_64) PWLIB_MACHTYPE=x86_64 ;
|
||||
P_64BIT=1 ;
|
||||
LIB64=1 ;
|
||||
;;
|
||||
|
||||
alpha | alphaev56 | alphaev6 | alphaev67 | alphaev7) PWLIB_MACHTYPE=alpha ;
|
||||
P_64BIT=1 ;
|
||||
;;
|
||||
|
||||
sparc ) PWLIB_MACHTYPE=sparc ;
|
||||
;;
|
||||
|
||||
powerpc ) PWLIB_MACHTYPE=ppc ;
|
||||
;;
|
||||
|
||||
ppc ) PWLIB_MACHTYPE=ppc ;
|
||||
;;
|
||||
|
||||
powerpc64 ) PWLIB_MACHTYPE=ppc64 ;
|
||||
P_64BIT=1 ;
|
||||
LIB64=1 ;
|
||||
;;
|
||||
|
||||
ppc64 ) PWLIB_MACHTYPE=ppc64 ;
|
||||
P_64BIT=1 ;
|
||||
LIB64=1 ;
|
||||
;;
|
||||
|
||||
ia64) PWLIB_MACHTYPE=ia64 ;
|
||||
P_64BIT=1 ;
|
||||
;;
|
||||
|
||||
s390x) PWLIB_MACHTYPE=s390x ;
|
||||
P_64BIT=1 ;
|
||||
LIB64=1 ;
|
||||
;;
|
||||
|
||||
s390) PWLIB_MACHTYPE=s390 ;
|
||||
;;
|
||||
|
||||
* ) PWLIB_MACHTYPE="$host_cpu";
|
||||
AC_MSG_WARN("CPU $PWLIB_MACHTYPE not recognized - proceed with caution!") ;;
|
||||
esac
|
||||
|
||||
PWLIB_PLATFORM="${PWLIB_OSTYPE}_${PWLIB_MACHTYPE}"
|
||||
|
||||
AC_SUBST([PWLIB_PLATFORM])
|
||||
])
|
||||
|
||||
|
||||
AC_DEFUN(
|
||||
[AST_CHECK_OPENH323], [
|
||||
OPENH323_INCDIR=
|
||||
OPENH323_LIBDIR=
|
||||
if test "${OPENH323DIR:-unset}" != "unset" ; then
|
||||
AC_CHECK_FILE(${OPENH323DIR}/version.h, HAS_OPENH323=1, )
|
||||
fi
|
||||
if test "${HAS_OPENH323:-unset}" = "unset" ; then
|
||||
AC_CHECK_FILE(${PWLIBDIR}/../openh323/version.h, OPENH323DIR="${PWLIBDIR}/../openh323"; HAS_OPENH323=1, )
|
||||
if test "${HAS_OPENH323:-unset}" != "unset" ; then
|
||||
OPENH323DIR="${PWLIBDIR}/../openh323"
|
||||
AC_CHECK_FILE(${OPENH323DIR}/include/h323.h, , OPENH323_INCDIR="${PWLIB_INCDIR}/openh323"; OPENH323_LIBDIR="${PWLIB_LIBDIR}")
|
||||
else
|
||||
AC_CHECK_FILE(${HOME}/openh323/include/h323.h, HAS_OPENH323=1, )
|
||||
if test "${HAS_OPENH323:-unset}" != "unset" ; then
|
||||
OPENH323DIR="${HOME}/openh323"
|
||||
else
|
||||
AC_CHECK_FILE(/usr/local/include/openh323/h323.h, HAS_OPENH323=1, )
|
||||
if test "${HAS_OPENH323:-unset}" != "unset" ; then
|
||||
OPENH323DIR="/usr/local/share/openh323"
|
||||
OPENH323_INCDIR="/usr/local/include/openh323"
|
||||
if test "x$LIB64" != "x"; then
|
||||
OPENH323_LIBDIR="/usr/local/lib64"
|
||||
else
|
||||
OPENH323_LIBDIR="/usr/local/lib"
|
||||
fi
|
||||
else
|
||||
AC_CHECK_FILE(/usr/include/openh323/h323.h, HAS_OPENH323=1, )
|
||||
if test "${HAS_OPENH323:-unset}" != "unset" ; then
|
||||
OPENH323DIR="/usr/share/openh323"
|
||||
OPENH323_INCDIR="/usr/include/openh323"
|
||||
if test "x$LIB64" != "x"; then
|
||||
OPENH323_LIBDIR="/usr/lib64"
|
||||
else
|
||||
OPENH323_LIBDIR="/usr/lib"
|
||||
fi
|
||||
fi
|
||||
fi
|
||||
fi
|
||||
fi
|
||||
fi
|
||||
|
||||
if test "${HAS_OPENH323:-unset}" != "unset" ; then
|
||||
if test "${OPENH323_INCDIR:-unset}" = "unset"; then
|
||||
OPENH323_INCDIR="${OPENH323DIR}/include"
|
||||
fi
|
||||
if test "${OPENH323_LIBDIR:-unset}" = "unset"; then
|
||||
OPENH323_LIBDIR="${OPENH323DIR}/lib"
|
||||
fi
|
||||
|
||||
OPENH323_LIBDIR="`cd ${OPENH323_LIBDIR}; pwd`"
|
||||
OPENH323_INCDIR="`cd ${OPENH323_INCDIR}; pwd`"
|
||||
OPENH323DIR="`cd ${OPENH323DIR}; pwd`"
|
||||
|
||||
AC_SUBST([OPENH323DIR])
|
||||
AC_SUBST([OPENH323_INCDIR])
|
||||
AC_SUBST([OPENH323_LIBDIR])
|
||||
fi
|
||||
])
|
||||
|
||||
|
||||
AC_DEFUN(
|
||||
[AST_CHECK_PWLIB_VERSION], [
|
||||
if test "${HAS_$2:-unset}" != "unset"; then
|
||||
$2_VERSION=`grep "$2_VERSION" ${$2_INCDIR}/$3 | cut -f2 -d ' ' | sed -e 's/"//g'`
|
||||
$2_MAJOR_VERSION=`echo ${$2_VERSION} | cut -f1 -d.`
|
||||
$2_MINOR_VERSION=`echo ${$2_VERSION} | cut -f2 -d.`
|
||||
$2_BUILD_NUMBER=`echo ${$2_VERSION} | cut -f3 -d.`
|
||||
let $2_VER=${$2_MAJOR_VERSION}*10000+${$2_MINOR_VERSION}*100+${$2_BUILD_NUMBER}
|
||||
let $2_REQ=$4*10000+$5*100+$6
|
||||
|
||||
AC_MSG_CHECKING(if $1 version ${$2_VERSION} is compatible with chan_h323)
|
||||
if test ${$2_VER} -lt ${$2_REQ}; then
|
||||
AC_MSG_RESULT(no)
|
||||
unset HAS_$2
|
||||
else
|
||||
AC_MSG_RESULT(yes)
|
||||
fi
|
||||
fi
|
||||
])
|
||||
|
||||
|
||||
AC_DEFUN(
|
||||
[AST_CHECK_PWLIB_BUILD], [
|
||||
if test "${HAS_$2:-unset}" != "unset"; then
|
||||
AC_MSG_CHECKING($1 installation validity)
|
||||
|
||||
saved_cppflags="${CPPFLAGS}"
|
||||
saved_libs="${LIBS}"
|
||||
if test "${$2_LIB:-unset}" != "unset"; then
|
||||
LIBS="${LIBS} ${$2_LIB} $7"
|
||||
else
|
||||
LIBS="${LIBS} -L${$2_LIBDIR} -l${PLATFORM_$2} $7"
|
||||
fi
|
||||
CPPFLAGS="${CPPFLAGS} -I${$2_INCDIR} $6"
|
||||
|
||||
AC_LANG_PUSH([C++])
|
||||
|
||||
AC_LINK_IFELSE(
|
||||
[AC_LANG_PROGRAM([$4],[$5])],
|
||||
[ AC_MSG_RESULT(yes)
|
||||
ac_cv_lib_$2="yes"
|
||||
],
|
||||
[ AC_MSG_RESULT(no)
|
||||
ac_cv_lib_$2="no"
|
||||
]
|
||||
)
|
||||
|
||||
AC_LANG_POP([C++])
|
||||
|
||||
LIBS="${saved_libs}"
|
||||
CPPFLAGS="${saved_cppflags}"
|
||||
|
||||
if test "${ac_cv_lib_$2}" = "yes"; then
|
||||
if test "${$2_LIB:-undef}" = "undef"; then
|
||||
if test "${$2_LIBDIR}" != "" -a "${$2_LIBDIR}" != "/usr/lib"; then
|
||||
$2_LIB="-L${$2_LIBDIR} -l${PLATFORM_$2}"
|
||||
else
|
||||
$2_LIB="-l${PLATFORM_$2}"
|
||||
fi
|
||||
fi
|
||||
if test "${$2_INCDIR}" != "" -a "${$2_INCDIR}" != "/usr/include"; then
|
||||
$2_INCLUDE="-I${$2_INCDIR}"
|
||||
fi
|
||||
PBX_$2=1
|
||||
AC_DEFINE([HAVE_$2], 1, [$3])
|
||||
fi
|
||||
fi
|
||||
])
|
||||
|
||||
AC_DEFUN(
|
||||
[AST_CHECK_OPENH323_BUILD], [
|
||||
if test "${HAS_OPENH323:-unset}" != "unset"; then
|
||||
AC_MSG_CHECKING(OpenH323 build option)
|
||||
OPENH323_SUFFIX=
|
||||
prefixes="h323_${PWLIB_PLATFORM}_ h323_ openh323"
|
||||
for pfx in $prefixes; do
|
||||
files=`ls -l ${OPENH323_LIBDIR}/lib${pfx}*.so* 2>/dev/null`
|
||||
libfile=
|
||||
if test -n "$files"; then
|
||||
for f in $files; do
|
||||
if test -f $f -a ! -L $f; then
|
||||
libfile=`basename $f`
|
||||
break;
|
||||
fi
|
||||
done
|
||||
fi
|
||||
if test -n "$libfile"; then
|
||||
OPENH323_PREFIX=$pfx
|
||||
break;
|
||||
fi
|
||||
done
|
||||
if test "${libfile:-unset}" != "unset"; then
|
||||
OPENH323_SUFFIX=`eval "echo ${libfile} | sed -e 's/lib${OPENH323_PREFIX}\(@<:@^.@:>@*\)\..*/\1/'"`
|
||||
fi
|
||||
case "${OPENH323_SUFFIX}" in
|
||||
n)
|
||||
OPENH323_BUILD="notrace";;
|
||||
r)
|
||||
OPENH323_BUILD="opt";;
|
||||
d)
|
||||
OPENH323_BUILD="debug";;
|
||||
*)
|
||||
if test "${OPENH323_PREFIX:-undef}" = "openh323"; then
|
||||
notrace=`eval "grep NOTRACE ${OPENH323DIR}/openh323u.mak | grep = | sed -e 's/@<:@A-Z0-9_@:>@*@<:@ @:>@*=@<:@ @:>@*//'"`
|
||||
if test "x$notrace" = "x"; then
|
||||
notrace="0"
|
||||
fi
|
||||
if test "$notrace" -ne 0; then
|
||||
OPENH323_BUILD="notrace"
|
||||
else
|
||||
OPENH323_BUILD="opt"
|
||||
fi
|
||||
OPENH323_LIB="-l${OPENH323_PREFIX}"
|
||||
else
|
||||
OPENH323_BUILD="notrace"
|
||||
fi
|
||||
;;
|
||||
esac
|
||||
AC_MSG_RESULT(${OPENH323_BUILD})
|
||||
|
||||
AC_SUBST([OPENH323_SUFFIX])
|
||||
AC_SUBST([OPENH323_BUILD])
|
||||
fi
|
||||
])
|
||||
|
||||
|
||||
# AST_FUNC_FORK
|
||||
# -------------
|
||||
AN_FUNCTION([fork], [AST_FUNC_FORK])
|
||||
AN_FUNCTION([vfork], [AST_FUNC_FORK])
|
||||
AC_DEFUN([AST_FUNC_FORK],
|
||||
[AC_REQUIRE([AC_TYPE_PID_T])dnl
|
||||
AC_CHECK_HEADERS(vfork.h)
|
||||
AC_CHECK_FUNCS(fork vfork)
|
||||
if test "x$ac_cv_func_fork" = xyes; then
|
||||
_AST_FUNC_FORK
|
||||
else
|
||||
ac_cv_func_fork_works=$ac_cv_func_fork
|
||||
fi
|
||||
if test "x$ac_cv_func_fork_works" = xcross; then
|
||||
case $host in
|
||||
*-*-amigaos* | *-*-msdosdjgpp* | *-*-uclinux* | *-*-linux-uclibc* )
|
||||
# Override, as these systems have only a dummy fork() stub
|
||||
ac_cv_func_fork_works=no
|
||||
;;
|
||||
*)
|
||||
ac_cv_func_fork_works=yes
|
||||
;;
|
||||
esac
|
||||
AC_MSG_WARN([result $ac_cv_func_fork_works guessed because of cross compilation])
|
||||
fi
|
||||
ac_cv_func_vfork_works=$ac_cv_func_vfork
|
||||
if test "x$ac_cv_func_vfork" = xyes; then
|
||||
_AC_FUNC_VFORK
|
||||
fi;
|
||||
if test "x$ac_cv_func_fork_works" = xcross; then
|
||||
ac_cv_func_vfork_works=$ac_cv_func_vfork
|
||||
AC_MSG_WARN([result $ac_cv_func_vfork_works guessed because of cross compilation])
|
||||
fi
|
||||
|
||||
if test "x$ac_cv_func_vfork_works" = xyes; then
|
||||
AC_DEFINE(HAVE_WORKING_VFORK, 1, [Define to 1 if `vfork' works.])
|
||||
else
|
||||
AC_DEFINE(vfork, fork, [Define as `fork' if `vfork' does not work.])
|
||||
fi
|
||||
if test "x$ac_cv_func_fork_works" = xyes; then
|
||||
AC_DEFINE(HAVE_WORKING_FORK, 1, [Define to 1 if `fork' works.])
|
||||
fi
|
||||
])# AST_FUNC_FORK
|
||||
|
||||
|
||||
# _AST_FUNC_FORK
|
||||
# -------------
|
||||
AC_DEFUN([_AST_FUNC_FORK],
|
||||
[AC_CACHE_CHECK(for working fork, ac_cv_func_fork_works,
|
||||
[AC_RUN_IFELSE(
|
||||
[AC_LANG_PROGRAM([AC_INCLUDES_DEFAULT],
|
||||
[
|
||||
/* By Ruediger Kuhlmann. */
|
||||
return fork () < 0;
|
||||
])],
|
||||
[ac_cv_func_fork_works=yes],
|
||||
[ac_cv_func_fork_works=no],
|
||||
[ac_cv_func_fork_works=cross])])]
|
||||
)# _AST_FUNC_FORK
|
||||
|
||||
# AST_PROG_LD
|
||||
# ----------
|
||||
# find the pathname to the GNU or non-GNU linker
|
||||
AC_DEFUN([AST_PROG_LD],
|
||||
[AC_ARG_WITH([gnu-ld],
|
||||
[AC_HELP_STRING([--with-gnu-ld],
|
||||
[assume the C compiler uses GNU ld @<:@default=no@:>@])],
|
||||
[test "$withval" = no || with_gnu_ld=yes],
|
||||
[with_gnu_ld=no])
|
||||
AC_REQUIRE([AST_PROG_SED])dnl
|
||||
AC_REQUIRE([AC_PROG_CC])dnl
|
||||
AC_REQUIRE([AC_CANONICAL_HOST])dnl
|
||||
AC_REQUIRE([AC_CANONICAL_BUILD])dnl
|
||||
ac_prog=ld
|
||||
if test "$GCC" = yes; then
|
||||
# Check if gcc -print-prog-name=ld gives a path.
|
||||
AC_MSG_CHECKING([for ld used by $CC])
|
||||
case $host in
|
||||
*-*-mingw*)
|
||||
# gcc leaves a trailing carriage return which upsets mingw
|
||||
ac_prog=`($CC -print-prog-name=ld) 2>&5 | tr -d '\015'` ;;
|
||||
*)
|
||||
ac_prog=`($CC -print-prog-name=ld) 2>&5` ;;
|
||||
esac
|
||||
case $ac_prog in
|
||||
# Accept absolute paths.
|
||||
[[\\/]]* | ?:[[\\/]]*)
|
||||
re_direlt='/[[^/]][[^/]]*/\.\./'
|
||||
# Canonicalize the pathname of ld
|
||||
ac_prog=`echo $ac_prog| $SED 's%\\\\%/%g'`
|
||||
while echo $ac_prog | grep "$re_direlt" > /dev/null 2>&1; do
|
||||
ac_prog=`echo $ac_prog| $SED "s%$re_direlt%/%"`
|
||||
done
|
||||
test -z "$LD" && LD="$ac_prog"
|
||||
;;
|
||||
"")
|
||||
# If it fails, then pretend we aren't using GCC.
|
||||
ac_prog=ld
|
||||
;;
|
||||
*)
|
||||
# If it is relative, then search for the first ld in PATH.
|
||||
with_gnu_ld=unknown
|
||||
;;
|
||||
esac
|
||||
elif test "$with_gnu_ld" = yes; then
|
||||
AC_MSG_CHECKING([for GNU ld])
|
||||
else
|
||||
AC_MSG_CHECKING([for non-GNU ld])
|
||||
fi
|
||||
AC_CACHE_VAL(lt_cv_path_LD,
|
||||
[if test -z "$LD"; then
|
||||
lt_save_ifs="$IFS"; IFS=$PATH_SEPARATOR
|
||||
for ac_dir in $PATH; do
|
||||
IFS="$lt_save_ifs"
|
||||
test -z "$ac_dir" && ac_dir=.
|
||||
if test -f "$ac_dir/$ac_prog" || test -f "$ac_dir/$ac_prog$ac_exeext"; then
|
||||
lt_cv_path_LD="$ac_dir/$ac_prog"
|
||||
# Check to see if the program is GNU ld. I'd rather use --version,
|
||||
# but apparently some variants of GNU ld only accept -v.
|
||||
# Break only if it was the GNU/non-GNU ld that we prefer.
|
||||
case `"$lt_cv_path_LD" -v 2>&1 </dev/null` in
|
||||
*GNU* | *'with BFD'*)
|
||||
test "$with_gnu_ld" != no && break
|
||||
;;
|
||||
*)
|
||||
test "$with_gnu_ld" != yes && break
|
||||
;;
|
||||
esac
|
||||
fi
|
||||
done
|
||||
IFS="$lt_save_ifs"
|
||||
else
|
||||
lt_cv_path_LD="$LD" # Let the user override the test with a path.
|
||||
fi])
|
||||
LD="$lt_cv_path_LD"
|
||||
if test -n "$LD"; then
|
||||
AC_MSG_RESULT($LD)
|
||||
else
|
||||
AC_MSG_RESULT(no)
|
||||
fi
|
||||
test -z "$LD" && AC_MSG_ERROR([no acceptable ld found in \$PATH])
|
||||
AST_PROG_LD_GNU
|
||||
])# AST_PROG_LD
|
||||
|
||||
|
||||
# AST_PROG_LD_GNU
|
||||
# --------------
|
||||
AC_DEFUN([AST_PROG_LD_GNU],
|
||||
[AC_REQUIRE([AST_PROG_EGREP])dnl
|
||||
AC_CACHE_CHECK([if the linker ($LD) is GNU ld], lt_cv_prog_gnu_ld,
|
||||
[# I'd rather use --version here, but apparently some GNU lds only accept -v.
|
||||
case `$LD -v 2>&1 </dev/null` in
|
||||
*GNU* | *'with BFD'*)
|
||||
lt_cv_prog_gnu_ld=yes
|
||||
;;
|
||||
*)
|
||||
lt_cv_prog_gnu_ld=no
|
||||
;;
|
||||
esac])
|
||||
with_gnu_ld=$lt_cv_prog_gnu_ld
|
||||
])# AST_PROG_LD_GNU
|
||||
|
||||
# AST_PROG_EGREP
|
||||
# -------------
|
||||
m4_ifndef([AST_PROG_EGREP], [AC_DEFUN([AST_PROG_EGREP],
|
||||
[AC_CACHE_CHECK([for egrep], [ac_cv_prog_egrep],
|
||||
[if echo a | (grep -E '(a|b)') >/dev/null 2>&1
|
||||
then ac_cv_prog_egrep='grep -E'
|
||||
else ac_cv_prog_egrep='egrep'
|
||||
fi])
|
||||
EGREP=$ac_cv_prog_egrep
|
||||
AC_SUBST([EGREP])
|
||||
])]) # AST_PROG_EGREP
|
||||
|
||||
# AST_PROG_SED
|
||||
# -----------
|
||||
# Check for a fully functional sed program that truncates
|
||||
# as few characters as possible. Prefer GNU sed if found.
|
||||
AC_DEFUN([AST_PROG_SED],
|
||||
[AC_CACHE_CHECK([for a sed that does not truncate output], ac_cv_path_SED,
|
||||
[dnl ac_script should not contain more than 99 commands (for HP-UX sed),
|
||||
dnl but more than about 7000 bytes, to catch a limit in Solaris 8 /usr/ucb/sed.
|
||||
ac_script=s/aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/bbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbb/
|
||||
for ac_i in 1 2 3 4 5 6 7; do
|
||||
ac_script="$ac_script$as_nl$ac_script"
|
||||
done
|
||||
echo "$ac_script" | sed 99q >conftest.sed
|
||||
$as_unset ac_script || ac_script=
|
||||
_AC_PATH_PROG_FEATURE_CHECK(SED, [sed gsed],
|
||||
[_AC_FEATURE_CHECK_LENGTH([ac_path_SED], [ac_cv_path_SED],
|
||||
["$ac_path_SED" -f conftest.sed])])])
|
||||
SED="$ac_cv_path_SED"
|
||||
AC_SUBST([SED])dnl
|
||||
rm -f conftest.sed
|
||||
])# AST_PROG_SED
|
||||
|
||||
dnl @synopsis ACX_PTHREAD([ACTION-IF-FOUND[, ACTION-IF-NOT-FOUND]])
|
||||
dnl
|
||||
dnl @summary figure out how to build C programs using POSIX threads
|
||||
dnl
|
||||
dnl This macro figures out how to build C programs using POSIX threads.
|
||||
dnl It sets the PTHREAD_LIBS output variable to the threads library and
|
||||
dnl linker flags, and the PTHREAD_CFLAGS output variable to any special
|
||||
dnl C compiler flags that are needed. (The user can also force certain
|
||||
dnl compiler flags/libs to be tested by setting these environment
|
||||
dnl variables.)
|
||||
dnl
|
||||
dnl Also sets PTHREAD_CC to any special C compiler that is needed for
|
||||
dnl multi-threaded programs (defaults to the value of CC otherwise).
|
||||
dnl (This is necessary on AIX to use the special cc_r compiler alias.)
|
||||
dnl
|
||||
dnl NOTE: You are assumed to not only compile your program with these
|
||||
dnl flags, but also link it with them as well. e.g. you should link
|
||||
dnl with $PTHREAD_CC $CFLAGS $PTHREAD_CFLAGS $LDFLAGS ... $PTHREAD_LIBS
|
||||
dnl $LIBS
|
||||
dnl
|
||||
dnl If you are only building threads programs, you may wish to use
|
||||
dnl these variables in your default LIBS, CFLAGS, and CC:
|
||||
dnl
|
||||
dnl LIBS="$PTHREAD_LIBS $LIBS"
|
||||
dnl CFLAGS="$CFLAGS $PTHREAD_CFLAGS"
|
||||
dnl CC="$PTHREAD_CC"
|
||||
dnl
|
||||
dnl In addition, if the PTHREAD_CREATE_JOINABLE thread-attribute
|
||||
dnl constant has a nonstandard name, defines PTHREAD_CREATE_JOINABLE to
|
||||
dnl that name (e.g. PTHREAD_CREATE_UNDETACHED on AIX).
|
||||
dnl
|
||||
dnl ACTION-IF-FOUND is a list of shell commands to run if a threads
|
||||
dnl library is found, and ACTION-IF-NOT-FOUND is a list of commands to
|
||||
dnl run it if it is not found. If ACTION-IF-FOUND is not specified, the
|
||||
dnl default action will define HAVE_PTHREAD.
|
||||
dnl
|
||||
dnl Please let the authors know if this macro fails on any platform, or
|
||||
dnl if you have any other suggestions or comments. This macro was based
|
||||
dnl on work by SGJ on autoconf scripts for FFTW (www.fftw.org) (with
|
||||
dnl help from M. Frigo), as well as ac_pthread and hb_pthread macros
|
||||
dnl posted by Alejandro Forero Cuervo to the autoconf macro repository.
|
||||
dnl We are also grateful for the helpful feedback of numerous users.
|
||||
dnl
|
||||
dnl @category InstalledPackages
|
||||
dnl @author Steven G. Johnson <stevenj@alum.mit.edu>
|
||||
dnl @version 2006-05-29
|
||||
dnl @license GPLWithACException
|
||||
|
||||
AC_DEFUN([ACX_PTHREAD], [
|
||||
AC_REQUIRE([AC_CANONICAL_HOST])
|
||||
AC_LANG_SAVE
|
||||
AC_LANG_C
|
||||
acx_pthread_ok=no
|
||||
|
||||
# We used to check for pthread.h first, but this fails if pthread.h
|
||||
# requires special compiler flags (e.g. on True64 or Sequent).
|
||||
# It gets checked for in the link test anyway.
|
||||
|
||||
# First of all, check if the user has set any of the PTHREAD_LIBS,
|
||||
# etcetera environment variables, and if threads linking works using
|
||||
# them:
|
||||
if test x"$PTHREAD_LIBS$PTHREAD_CFLAGS" != x; then
|
||||
save_CFLAGS="$CFLAGS"
|
||||
CFLAGS="$CFLAGS $PTHREAD_CFLAGS"
|
||||
save_LIBS="$LIBS"
|
||||
LIBS="$PTHREAD_LIBS $LIBS"
|
||||
AC_MSG_CHECKING([for pthread_join in LIBS=$PTHREAD_LIBS with CFLAGS=$PTHREAD_CFLAGS])
|
||||
AC_TRY_LINK_FUNC(pthread_join, acx_pthread_ok=yes)
|
||||
AC_MSG_RESULT($acx_pthread_ok)
|
||||
if test x"$acx_pthread_ok" = xno; then
|
||||
PTHREAD_LIBS=""
|
||||
PTHREAD_CFLAGS=""
|
||||
fi
|
||||
LIBS="$save_LIBS"
|
||||
CFLAGS="$save_CFLAGS"
|
||||
fi
|
||||
|
||||
# We must check for the threads library under a number of different
|
||||
# names; the ordering is very important because some systems
|
||||
# (e.g. DEC) have both -lpthread and -lpthreads, where one of the
|
||||
# libraries is broken (non-POSIX).
|
||||
|
||||
# Create a list of thread flags to try. Items starting with a "-" are
|
||||
# C compiler flags, and other items are library names, except for "none"
|
||||
# which indicates that we try without any flags at all, and "pthread-config"
|
||||
# which is a program returning the flags for the Pth emulation library.
|
||||
|
||||
acx_pthread_flags="pthreads none -Kthread -kthread lthread -pthread -pthreads -mthreads pthread --thread-safe -mt pthread-config"
|
||||
|
||||
# The ordering *is* (sometimes) important. Some notes on the
|
||||
# individual items follow:
|
||||
|
||||
# pthreads: AIX (must check this before -lpthread)
|
||||
# none: in case threads are in libc; should be tried before -Kthread and
|
||||
# other compiler flags to prevent continual compiler warnings
|
||||
# -Kthread: Sequent (threads in libc, but -Kthread needed for pthread.h)
|
||||
# -kthread: FreeBSD kernel threads (preferred to -pthread since SMP-able)
|
||||
# lthread: LinuxThreads port on FreeBSD (also preferred to -pthread)
|
||||
# -pthread: Linux/gcc (kernel threads), BSD/gcc (userland threads)
|
||||
# -pthreads: Solaris/gcc
|
||||
# -mthreads: Mingw32/gcc, Lynx/gcc
|
||||
# -mt: Sun Workshop C (may only link SunOS threads [-lthread], but it
|
||||
# doesn't hurt to check since this sometimes defines pthreads too;
|
||||
# also defines -D_REENTRANT)
|
||||
# ... -mt is also the pthreads flag for HP/aCC
|
||||
# pthread: Linux, etcetera
|
||||
# --thread-safe: KAI C++
|
||||
# pthread-config: use pthread-config program (for GNU Pth library)
|
||||
|
||||
case "${host_cpu}-${host_os}" in
|
||||
*solaris*)
|
||||
|
||||
# On Solaris (at least, for some versions), libc contains stubbed
|
||||
# (non-functional) versions of the pthreads routines, so link-based
|
||||
# tests will erroneously succeed. (We need to link with -pthreads/-mt/
|
||||
# -lpthread.) (The stubs are missing pthread_cleanup_push, or rather
|
||||
# a function called by this macro, so we could check for that, but
|
||||
# who knows whether they'll stub that too in a future libc.) So,
|
||||
# we'll just look for -pthreads and -lpthread first:
|
||||
|
||||
acx_pthread_flags="-pthreads pthread -mt -pthread $acx_pthread_flags"
|
||||
;;
|
||||
esac
|
||||
|
||||
if test x"$acx_pthread_ok" = xno; then
|
||||
for flag in $acx_pthread_flags; do
|
||||
|
||||
case $flag in
|
||||
none)
|
||||
AC_MSG_CHECKING([whether pthreads work without any flags])
|
||||
;;
|
||||
|
||||
-*)
|
||||
AC_MSG_CHECKING([whether pthreads work with $flag])
|
||||
PTHREAD_CFLAGS="$flag"
|
||||
;;
|
||||
|
||||
pthread-config)
|
||||
AC_CHECK_PROG(acx_pthread_config, pthread-config, yes, no)
|
||||
if test x"$acx_pthread_config" = xno; then continue; fi
|
||||
PTHREAD_CFLAGS="`pthread-config --cflags`"
|
||||
PTHREAD_LIBS="`pthread-config --ldflags` `pthread-config --libs`"
|
||||
;;
|
||||
|
||||
*)
|
||||
AC_MSG_CHECKING([for the pthreads library -l$flag])
|
||||
PTHREAD_LIBS="-l$flag"
|
||||
;;
|
||||
esac
|
||||
|
||||
save_LIBS="$LIBS"
|
||||
save_CFLAGS="$CFLAGS"
|
||||
LIBS="$PTHREAD_LIBS $LIBS"
|
||||
CFLAGS="$CFLAGS $PTHREAD_CFLAGS"
|
||||
|
||||
# Check for various functions. We must include pthread.h,
|
||||
# since some functions may be macros. (On the Sequent, we
|
||||
# need a special flag -Kthread to make this header compile.)
|
||||
# We check for pthread_join because it is in -lpthread on IRIX
|
||||
# while pthread_create is in libc. We check for pthread_attr_init
|
||||
# due to DEC craziness with -lpthreads. We check for
|
||||
# pthread_cleanup_push because it is one of the few pthread
|
||||
# functions on Solaris that doesn't have a non-functional libc stub.
|
||||
# We try pthread_create on general principles.
|
||||
AC_TRY_LINK([#include <pthread.h>],
|
||||
[pthread_t th; pthread_join(th, 0);
|
||||
pthread_attr_init(0); pthread_cleanup_push(0, 0);
|
||||
pthread_create(0,0,0,0); pthread_cleanup_pop(0); ],
|
||||
[acx_pthread_ok=yes])
|
||||
|
||||
LIBS="$save_LIBS"
|
||||
CFLAGS="$save_CFLAGS"
|
||||
|
||||
AC_MSG_RESULT($acx_pthread_ok)
|
||||
if test "x$acx_pthread_ok" = xyes; then
|
||||
break;
|
||||
fi
|
||||
|
||||
PTHREAD_LIBS=""
|
||||
PTHREAD_CFLAGS=""
|
||||
done
|
||||
fi
|
||||
|
||||
# Various other checks:
|
||||
if test "x$acx_pthread_ok" = xyes; then
|
||||
save_LIBS="$LIBS"
|
||||
LIBS="$PTHREAD_LIBS $LIBS"
|
||||
save_CFLAGS="$CFLAGS"
|
||||
CFLAGS="$CFLAGS $PTHREAD_CFLAGS"
|
||||
|
||||
# Detect AIX lossage: JOINABLE attribute is called UNDETACHED.
|
||||
AC_MSG_CHECKING([for joinable pthread attribute])
|
||||
attr_name=unknown
|
||||
for attr in PTHREAD_CREATE_JOINABLE PTHREAD_CREATE_UNDETACHED; do
|
||||
AC_TRY_LINK([#include <pthread.h>], [int attr=$attr; return attr;],
|
||||
[attr_name=$attr; break])
|
||||
done
|
||||
AC_MSG_RESULT($attr_name)
|
||||
if test "$attr_name" != PTHREAD_CREATE_JOINABLE; then
|
||||
AC_DEFINE_UNQUOTED(PTHREAD_CREATE_JOINABLE, $attr_name,
|
||||
[Define to necessary symbol if this constant
|
||||
uses a non-standard name on your system.])
|
||||
fi
|
||||
|
||||
AC_MSG_CHECKING([if more special flags are required for pthreads])
|
||||
flag=no
|
||||
case "${host_cpu}-${host_os}" in
|
||||
*-aix* | *-freebsd* | *-darwin*) flag="-D_THREAD_SAFE";;
|
||||
*solaris* | *-osf* | *-hpux*) flag="-D_REENTRANT";;
|
||||
esac
|
||||
AC_MSG_RESULT(${flag})
|
||||
if test "x$flag" != xno; then
|
||||
PTHREAD_CFLAGS="$flag $PTHREAD_CFLAGS"
|
||||
fi
|
||||
|
||||
LIBS="$save_LIBS"
|
||||
CFLAGS="$save_CFLAGS"
|
||||
|
||||
# More AIX lossage: must compile with xlc_r or cc_r
|
||||
if test x"$GCC" != xyes; then
|
||||
AC_CHECK_PROGS(PTHREAD_CC, xlc_r cc_r, ${CC})
|
||||
else
|
||||
PTHREAD_CC=$CC
|
||||
fi
|
||||
else
|
||||
PTHREAD_CC="$CC"
|
||||
fi
|
||||
|
||||
AC_SUBST(PTHREAD_LIBS)
|
||||
AC_SUBST(PTHREAD_CFLAGS)
|
||||
AC_SUBST(PTHREAD_CC)
|
||||
|
||||
# Finally, execute ACTION-IF-FOUND/ACTION-IF-NOT-FOUND:
|
||||
if test x"$acx_pthread_ok" = xyes; then
|
||||
ifelse([$1],,AC_DEFINE(HAVE_PTHREAD,1,[Define if you have POSIX threads libraries and header files.]),[$1])
|
||||
:
|
||||
else
|
||||
acx_pthread_ok=no
|
||||
$2
|
||||
fi
|
||||
AC_LANG_RESTORE
|
||||
])dnl ACX_PTHREAD
|
||||
20
addmailbox
Normal file
20
addmailbox
Normal file
@@ -0,0 +1,20 @@
|
||||
#!/bin/sh
|
||||
VMHOME=/var/spool/asterisk/vm
|
||||
SNDHOME=/var/lib/asterisk/sounds
|
||||
echo -n "Enter mailbox number: "
|
||||
read mailbox
|
||||
mkdir -p ${VMHOME}/${mailbox}
|
||||
mkdir -p ${VMHOME}/${mailbox}/INBOX
|
||||
cat ${SNDHOME}/vm-theperson.gsm > ${VMHOME}/${mailbox}/unavail.gsm
|
||||
cat ${SNDHOME}/vm-theperson.gsm > ${VMHOME}/${mailbox}/busy.gsm
|
||||
cat ${SNDHOME}/vm-extension.gsm > ${VMHOME}/${mailbox}/greet.gsm
|
||||
nums=`echo $mailbox | sed 's/./ \0/g'`
|
||||
for x in $nums; do
|
||||
cat ${SNDHOME}/digits/${x}.gsm >> ${VMHOME}/${mailbox}/unavail.gsm
|
||||
cat ${SNDHOME}/digits/${x}.gsm >> ${VMHOME}/${mailbox}/busy.gsm
|
||||
cat ${SNDHOME}/digits/${x}.gsm >> ${VMHOME}/${mailbox}/greet.gsm
|
||||
done
|
||||
|
||||
cat ${SNDHOME}/vm-isunavail.gsm >> ${VMHOME}/${mailbox}/unavail.gsm
|
||||
cat ${SNDHOME}/vm-isonphone.gsm >> ${VMHOME}/${mailbox}/busy.gsm
|
||||
|
||||
@@ -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);
|
||||
|
||||
38
agi/Makefile
38
agi/Makefile
@@ -1,47 +1,27 @@
|
||||
#
|
||||
# Asterisk -- A telephony toolkit for Linux.
|
||||
#
|
||||
# Makefile for AGI-related stuff
|
||||
# Makefile for PBX frontends (dynamically loaded)
|
||||
#
|
||||
# Copyright (C) 1999-2006, Digium
|
||||
# Copyright (C) 1999, Mark Spencer
|
||||
#
|
||||
# Mark Spencer <markster@digium.com>
|
||||
# Mark Spencer <markster@linux-support.net>
|
||||
#
|
||||
# This program is free software, distributed under the terms of
|
||||
# the GNU General Public License
|
||||
#
|
||||
|
||||
.PHONY: clean all uninstall
|
||||
AGIS=agi-test.agi
|
||||
|
||||
AGIS=agi-test.agi eagi-test eagi-sphinx-test jukebox.agi
|
||||
|
||||
ifeq ($(OSARCH),SunOS)
|
||||
LIBS+=-lsocket -lnsl
|
||||
endif
|
||||
|
||||
include $(ASTTOPDIR)/Makefile.rules
|
||||
CFLAGS+=
|
||||
|
||||
all: $(AGIS)
|
||||
|
||||
strcompat.c: ../main/strcompat.c
|
||||
@cp $< $@
|
||||
|
||||
eagi-test: eagi-test.o strcompat.o
|
||||
|
||||
eagi-sphinx-test: eagi-sphinx-test.o
|
||||
|
||||
install: all
|
||||
mkdir -p $(DESTDIR)$(AGI_DIR)
|
||||
for x in $(AGIS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(AGI_DIR) ; done
|
||||
|
||||
uninstall:
|
||||
for x in $(AGIS); do rm -f $(DESTDIR)$(AGI_DIR)/$$x ; done
|
||||
for x in $(AGIS); do $(INSTALL) -m 755 $$x $(AGI_DIR) ; done
|
||||
|
||||
clean:
|
||||
rm -f *.so *.o look eagi-test eagi-sphinx-test
|
||||
rm -f .*.o.d .*.oo.d
|
||||
rm -f strcompat.c
|
||||
rm -f *.so *.o look
|
||||
|
||||
ifneq ($(wildcard .*.d),)
|
||||
include .*.d
|
||||
endif
|
||||
%.so : %.o
|
||||
$(CC) -shared -Xlinker -x -o $@ $<
|
||||
|
||||
@@ -1,11 +1,5 @@
|
||||
#!/usr/bin/perl
|
||||
use strict;
|
||||
|
||||
$|=1;
|
||||
|
||||
# Setup some variables
|
||||
my %AGI; my $tests = 0; my $fail = 0; my $pass = 0;
|
||||
|
||||
while(<STDIN>) {
|
||||
chomp;
|
||||
last unless length($_);
|
||||
@@ -15,7 +9,7 @@ while(<STDIN>) {
|
||||
}
|
||||
|
||||
print STDERR "AGI Environment Dump:\n";
|
||||
foreach my $i (sort keys %AGI) {
|
||||
foreach $i (sort keys %AGI) {
|
||||
print STDERR " -- $i = $AGI{$i}\n";
|
||||
}
|
||||
|
||||
@@ -41,38 +35,38 @@ sub checkresult {
|
||||
|
||||
print STDERR "1. Testing 'sendfile'...";
|
||||
print "STREAM FILE beep \"\"\n";
|
||||
my $result = <STDIN>;
|
||||
&checkresult($result);
|
||||
$result = <STDIN>;
|
||||
checkresult($result);
|
||||
|
||||
print STDERR "2. Testing 'sendtext'...";
|
||||
print "SEND TEXT \"hello world\"\n";
|
||||
my $result = <STDIN>;
|
||||
&checkresult($result);
|
||||
$result = <STDIN>;
|
||||
checkresult($result);
|
||||
|
||||
print STDERR "3. Testing 'sendimage'...";
|
||||
print "SEND IMAGE asterisk-image\n";
|
||||
my $result = <STDIN>;
|
||||
&checkresult($result);
|
||||
$result = <STDIN>;
|
||||
checkresult($result);
|
||||
|
||||
print STDERR "4. Testing 'saynumber'...";
|
||||
print "SAY NUMBER 192837465 \"\"\n";
|
||||
my $result = <STDIN>;
|
||||
&checkresult($result);
|
||||
$result = <STDIN>;
|
||||
checkresult($result);
|
||||
|
||||
print STDERR "5. Testing 'waitdtmf'...";
|
||||
print "WAIT FOR DIGIT 1000\n";
|
||||
my $result = <STDIN>;
|
||||
&checkresult($result);
|
||||
$result = <STDIN>;
|
||||
checkresult($result);
|
||||
|
||||
print STDERR "6. Testing 'record'...";
|
||||
print "RECORD FILE testagi gsm 1234 3000\n";
|
||||
my $result = <STDIN>;
|
||||
&checkresult($result);
|
||||
$result = <STDIN>;
|
||||
checkresult($result);
|
||||
|
||||
print STDERR "6a. Testing 'record' playback...";
|
||||
print "STREAM FILE testagi \"\"\n";
|
||||
my $result = <STDIN>;
|
||||
&checkresult($result);
|
||||
$result = <STDIN>;
|
||||
checkresult($result);
|
||||
|
||||
print STDERR "================== Complete ======================\n";
|
||||
print STDERR "$tests tests completed, $pass passed, $fail failed\n";
|
||||
|
||||
@@ -1,222 +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.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);
|
||||
}
|
||||
165
agi/eagi-test.c
165
agi/eagi-test.c
@@ -1,165 +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.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);
|
||||
}
|
||||
@@ -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";
|
||||
}
|
||||
|
||||
488
agi/jukebox.agi
488
agi/jukebox.agi
@@ -1,488 +0,0 @@
|
||||
#!/usr/bin/perl
|
||||
#
|
||||
# Jukebox 0.2
|
||||
#
|
||||
# A music manager for Asterisk.
|
||||
#
|
||||
# Copyright (C) 2005-2006, Justin Tunney
|
||||
#
|
||||
# Justin Tunney <jesuscyborg@gmail.com>
|
||||
#
|
||||
# This program is free software, distributed under the terms of the
|
||||
# GNU General Public License v2.
|
||||
#
|
||||
# Keep it open source pigs
|
||||
#
|
||||
# --------------------------------------------------------------------
|
||||
#
|
||||
# Uses festival to list off all your MP3 music files over a channel in
|
||||
# a hierarchical fashion. Put this file in your agi-bin folder which
|
||||
# is located at: /var/lib/asterisk/agi-bin Be sure to chmod +x it!
|
||||
#
|
||||
# Invocation Example:
|
||||
# exten => 68742,1,Answer()
|
||||
# exten => 68742,2,agi,jukebox.agi|/home/justin/Music
|
||||
# exten => 68742,3,Hangup()
|
||||
#
|
||||
# exten => 68742,1,Answer()
|
||||
# exten => 68742,2,agi,jukebox.agi|/home/justin/Music|pm
|
||||
# exten => 68742,3,Hangup()
|
||||
#
|
||||
# Options:
|
||||
# p - Precache text2wave outputs for every possible filename.
|
||||
# It is much better to set this option because if a caller
|
||||
# presses a key during a cache operation, it will be ignored.
|
||||
# m - Go back to menu after playing song
|
||||
# g - Do not play the greeting message
|
||||
#
|
||||
# Usage Instructions:
|
||||
# - Press '*' to go up a directory. If you are in the root music
|
||||
# folder you will be exitted from the script.
|
||||
# - If you have a really long list of files, you can filter the list
|
||||
# at any time by pressing '#' and spelling out a few letters you
|
||||
# expect the files to start with. For example, if you wanted to
|
||||
# know what extension 'Requiem For A Dream' was, you'd type:
|
||||
# '#737'. Note, phone keypads don't include Q and Z. Q is 7 and
|
||||
# Z is 9.
|
||||
#
|
||||
# Notes:
|
||||
# - This AGI script uses the MP3Player command which uses the
|
||||
# mpg123 Program. Grab yourself a copy of this program by
|
||||
# going to http://www.mpg123.de/cgi-bin/sitexplorer.cgi?/mpg123/
|
||||
# Be sure to download mpg123-0.59r.tar.gz because it is known to
|
||||
# work with Asterisk and hopefully isn't the release with that
|
||||
# awful security problem. If you're using Fedora Core 3 with
|
||||
# Alsa like me, make linux-alsa isn't going to work. Do make
|
||||
# linux-devel and you're peachy keen.
|
||||
#
|
||||
# - You won't get nifty STDERR debug messages if you're using a
|
||||
# remote asterisk shell.
|
||||
#
|
||||
# - For some reason, caching certain files will generate the
|
||||
# error: 'using default diphone ax-ax for y-pau'. Example:
|
||||
# # echo "Depeche Mode - CUW - 05 - The Meaning of Love" | text2wave -o /var/jukeboxcache/jukeboxcache/Depeche_Mode/Depeche_Mode_-_CUW_-_05_-_The_Meaning_of_Love.mp3.ul -otype ulaw -
|
||||
# The temporary work around is to just touch these files.
|
||||
#
|
||||
# - The background app doesn't like to get more than 2031 chars
|
||||
# of input.
|
||||
#
|
||||
|
||||
use strict;
|
||||
|
||||
$|=1;
|
||||
|
||||
# Setup some variables
|
||||
my %AGI; my $tests = 0; my $fail = 0; my $pass = 0;
|
||||
my @masterCacheList = ();
|
||||
my $maxNumber = 10;
|
||||
|
||||
while (<STDIN>) {
|
||||
chomp;
|
||||
last unless length($_);
|
||||
if (/^agi_(\w+)\:\s+(.*)$/) {
|
||||
$AGI{$1} = $2;
|
||||
}
|
||||
}
|
||||
|
||||
# setup options
|
||||
my $SHOWGREET = 1;
|
||||
my $PRECACHE = 0;
|
||||
my $MENUAFTERSONG = 0;
|
||||
|
||||
$PRECACHE = 1 if $ARGV[1] =~ /p/;
|
||||
$MENUAFTERSONG = 1 if $ARGV[1] =~ /m/;
|
||||
$SHOWGREET = 0 if $ARGV[1] =~ /g/;
|
||||
|
||||
# setup folders
|
||||
my $MUSIC = $ARGV[0];
|
||||
$MUSIC = &rmts($MUSIC);
|
||||
my $FESTIVALCACHE = "/var/jukeboxcache";
|
||||
if (! -e $FESTIVALCACHE) {
|
||||
`mkdir -p -m0776 $FESTIVALCACHE`;
|
||||
}
|
||||
|
||||
# make sure we have some essential files
|
||||
if (! -e "$FESTIVALCACHE/jukebox_greet.ul") {
|
||||
`echo "Welcome to the Asterisk Jukebox" | text2wave -o $FESTIVALCACHE/jukebox_greet.ul -otype ulaw -`;
|
||||
}
|
||||
if (! -e "$FESTIVALCACHE/jukebox_press.ul") {
|
||||
`echo "Press" | text2wave -o $FESTIVALCACHE/jukebox_press.ul -otype ulaw -`;
|
||||
}
|
||||
if (! -e "$FESTIVALCACHE/jukebox_for.ul") {
|
||||
`echo "For" | text2wave -o $FESTIVALCACHE/jukebox_for.ul -otype ulaw -`;
|
||||
}
|
||||
if (! -e "$FESTIVALCACHE/jukebox_toplay.ul") {
|
||||
`echo "To play" | text2wave -o $FESTIVALCACHE/jukebox_toplay.ul -otype ulaw -`;
|
||||
}
|
||||
if (! -e "$FESTIVALCACHE/jukebox_nonefound.ul") {
|
||||
`echo "There were no music files found in this folder" | text2wave -o $FESTIVALCACHE/jukebox_nonefound.ul -otype ulaw -`;
|
||||
}
|
||||
if (! -e "$FESTIVALCACHE/jukebox_percent.ul") {
|
||||
`echo "Percent" | text2wave -o $FESTIVALCACHE/jukebox_percent.ul -otype ulaw -`;
|
||||
}
|
||||
if (! -e "$FESTIVALCACHE/jukebox_generate.ul") {
|
||||
`echo "Please wait while Astrisk Jukebox cashes the files of your music collection" | text2wave -o $FESTIVALCACHE/jukebox_generate.ul -otype ulaw -`;
|
||||
}
|
||||
if (! -e "$FESTIVALCACHE/jukebox_invalid.ul") {
|
||||
`echo "You have entered an invalid selection" | text2wave -o $FESTIVALCACHE/jukebox_invalid.ul -otype ulaw -`;
|
||||
}
|
||||
if (! -e "$FESTIVALCACHE/jukebox_thankyou.ul") {
|
||||
`echo "Thank you for using Astrisk Jukebox, Goodbye" | text2wave -o $FESTIVALCACHE/jukebox_thankyou.ul -otype ulaw -`;
|
||||
}
|
||||
|
||||
# greet the user
|
||||
if ($SHOWGREET) {
|
||||
print "EXEC Playback \"$FESTIVALCACHE/jukebox_greet\"\n";
|
||||
my $result = <STDIN>; &check_result($result);
|
||||
}
|
||||
|
||||
# go through the directories
|
||||
music_dir_cache() if $PRECACHE;
|
||||
music_dir_menu('/');
|
||||
|
||||
exit 0;
|
||||
|
||||
##########################################################################
|
||||
|
||||
sub music_dir_menu {
|
||||
my $dir = shift;
|
||||
|
||||
# generate a list of mp3's and directories and assign each one it's
|
||||
# own selection number. Then make sure that we've got a sound clip
|
||||
# for the file name
|
||||
if (!opendir(THEDIR, rmts($MUSIC.$dir))) {
|
||||
print STDERR "Failed to open music directory: $dir\n";
|
||||
exit 1;
|
||||
}
|
||||
my @files = sort readdir THEDIR;
|
||||
my $cnt = 1;
|
||||
my @masterBgList = ();
|
||||
|
||||
foreach my $file (@files) {
|
||||
chomp($file);
|
||||
if ($file ne '.' && $file ne '..' && $file ne 'festivalcache') { # ignore special files
|
||||
my $real_version = &rmts($MUSIC.$dir).'/'.$file;
|
||||
my $cache_version = &rmts($FESTIVALCACHE.$dir).'/'.$file.'.ul';
|
||||
my $cache_version2 = &rmts($FESTIVALCACHE.$dir).'/'.$file;
|
||||
my $cache_version_esc = &clean_file($cache_version);
|
||||
my $cache_version2_esc = &clean_file($cache_version2);
|
||||
|
||||
if (-d $real_version) {
|
||||
# 0:id 1:type 2:text2wav-file 3:for-filtering 4:the-directory 5:text2wav echo
|
||||
push(@masterBgList, [$cnt++, 1, $cache_version2_esc, &remove_special_chars($file), $file, "for the $file folder"]);
|
||||
} elsif ($real_version =~ /\.mp3$/) {
|
||||
# 0:id 1:type 2:text2wav-file 3:for-filtering 4:the-mp3
|
||||
push(@masterBgList, [$cnt++, 2, $cache_version2_esc, &remove_special_chars($file), $real_version, "to play $file"]);
|
||||
}
|
||||
}
|
||||
}
|
||||
close(THEDIR);
|
||||
|
||||
my @filterList = @masterBgList;
|
||||
|
||||
if (@filterList == 0) {
|
||||
print "EXEC Playback \"$FESTIVALCACHE/jukebox_nonefound\"\n";
|
||||
my $result = <STDIN>; &check_result($result);
|
||||
return 0;
|
||||
}
|
||||
|
||||
for (;;) {
|
||||
MYCONTINUE:
|
||||
|
||||
# play bg selections and figure out their selection
|
||||
my $digit = '';
|
||||
my $digitstr = '';
|
||||
for (my $n=0; $n<@filterList; $n++) {
|
||||
&cache_speech(&remove_file_extension($filterList[$n][5]), "$filterList[$n][2].ul") if ! -e "$filterList[$n][2].ul";
|
||||
&cache_speech("Press $filterList[$n][0]", "$FESTIVALCACHE/jukebox_$filterList[$n][0].ul") if ! -e "$FESTIVALCACHE/jukebox_$filterList[$n][0].ul";
|
||||
print "EXEC Background \"$filterList[$n][2]&$FESTIVALCACHE/jukebox_$filterList[$n][0]\"\n";
|
||||
my $result = <STDIN>;
|
||||
$digit = &check_result($result);
|
||||
if ($digit > 0) {
|
||||
$digitstr .= chr($digit);
|
||||
last;
|
||||
}
|
||||
}
|
||||
for (;;) {
|
||||
print "WAIT FOR DIGIT 3000\n";
|
||||
my $result = <STDIN>;
|
||||
$digit = &check_result($result);
|
||||
last if $digit <= 0;
|
||||
$digitstr .= chr($digit);
|
||||
}
|
||||
|
||||
# see if it's a valid selection
|
||||
print STDERR "Digits Entered: '$digitstr'\n";
|
||||
exit 0 if $digitstr eq '';
|
||||
my $found = 0;
|
||||
goto EXITSUB if $digitstr =~ /\*/;
|
||||
|
||||
# filter the list
|
||||
if ($digitstr =~ /^\#\d+/) {
|
||||
my $regexp = '';
|
||||
for (my $n=1; $n<length($digitstr); $n++) {
|
||||
my $d = substr($digitstr, $n, 1);
|
||||
if ($d == 2) {
|
||||
$regexp .= '[abc]';
|
||||
} elsif ($d == 3) {
|
||||
$regexp .= '[def]';
|
||||
} elsif ($d == 4) {
|
||||
$regexp .= '[ghi]';
|
||||
} elsif ($d == 5) {
|
||||
$regexp .= '[jkl]';
|
||||
} elsif ($d == 6) {
|
||||
$regexp .= '[mno]';
|
||||
} elsif ($d == 7) {
|
||||
$regexp .= '[pqrs]';
|
||||
} elsif ($d == 8) {
|
||||
$regexp .= '[tuv]';
|
||||
} elsif ($d == 9) {
|
||||
$regexp .= '[wxyz]';
|
||||
}
|
||||
}
|
||||
@filterList = ();
|
||||
for (my $n=1; $n<@masterBgList; $n++) {
|
||||
push(@filterList, $masterBgList[$n]) if $masterBgList[$n][3] =~ /^$regexp/i;
|
||||
}
|
||||
goto MYCONTINUE;
|
||||
}
|
||||
|
||||
for (my $n=0; $n<@masterBgList; $n++) {
|
||||
if ($digitstr == $masterBgList[$n][0]) {
|
||||
if ($masterBgList[$n][1] == 1) { # a folder
|
||||
&music_dir_menu(rmts($dir).'/'.$masterBgList[$n][4]);
|
||||
@filterList = @masterBgList;
|
||||
goto MYCONTINUE;
|
||||
} elsif ($masterBgList[$n][1] == 2) { # a file
|
||||
# because *'s scripting language is crunk and won't allow us to escape
|
||||
# funny filenames, we need to create a temporary symlink to the mp3
|
||||
# file
|
||||
my $mp3 = &escape_file($masterBgList[$n][4]);
|
||||
my $link = `mktemp`;
|
||||
chomp($link);
|
||||
$link .= '.mp3';
|
||||
print STDERR "ln -s $mp3 $link\n";
|
||||
my $cmdr = `ln -s $mp3 $link`;
|
||||
chomp($cmdr);
|
||||
print "Failed to create symlink to mp3: $cmdr\n" if $cmdr ne '';
|
||||
|
||||
print "EXEC MP3Player \"$link\"\n";
|
||||
my $result = <STDIN>; &check_result($result);
|
||||
|
||||
`rm $link`;
|
||||
|
||||
if (!$MENUAFTERSONG) {
|
||||
print "EXEC Playback \"$FESTIVALCACHE/jukebox_thankyou\"\n";
|
||||
my $result = <STDIN>; &check_result($result);
|
||||
exit 0;
|
||||
} else {
|
||||
goto MYCONTINUE;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
print "EXEC Playback \"$FESTIVALCACHE/jukebox_invalid\"\n";
|
||||
my $result = <STDIN>; &check_result($result);
|
||||
}
|
||||
EXITSUB:
|
||||
}
|
||||
|
||||
sub cache_speech {
|
||||
my $speech = shift;
|
||||
my $file = shift;
|
||||
|
||||
my $theDir = extract_file_dir($file);
|
||||
`mkdir -p -m0776 $theDir`;
|
||||
|
||||
print STDERR "echo \"$speech\" | text2wave -o $file -otype ulaw -\n";
|
||||
my $cmdr = `echo "$speech" | text2wave -o $file -otype ulaw -`;
|
||||
chomp($cmdr);
|
||||
if ($cmdr =~ /using default diphone/) {
|
||||
# temporary bug work around....
|
||||
`touch $file`;
|
||||
} elsif ($cmdr ne '') {
|
||||
print STDERR "Command Failed\n";
|
||||
exit 1;
|
||||
}
|
||||
}
|
||||
|
||||
sub music_dir_cache {
|
||||
# generate list of text2speech files to generate
|
||||
if (!music_dir_cache_genlist('/')) {
|
||||
print STDERR "Horrible Dreadful Error: No Music Found in $MUSIC!";
|
||||
exit 1;
|
||||
}
|
||||
|
||||
# add to list how many 'number' files we have to generate. We can't
|
||||
# use the SayNumber app in Asterisk because we want to chain all
|
||||
# talking in one Background command. We also want a consistent
|
||||
# voice...
|
||||
for (my $n=1; $n<=$maxNumber; $n++) {
|
||||
push(@masterCacheList, [3, "Press $n", "$FESTIVALCACHE/jukebox_$n.ul"]) if ! -e "$FESTIVALCACHE/jukebox_$n.ul";
|
||||
}
|
||||
|
||||
# now generate all these darn text2speech files
|
||||
if (@masterCacheList > 5) {
|
||||
print "EXEC Playback \"$FESTIVALCACHE/jukebox_generate\"\n";
|
||||
my $result = <STDIN>; &check_result($result);
|
||||
}
|
||||
my $theTime = time();
|
||||
for (my $n=0; $n < @masterCacheList; $n++) {
|
||||
my $cmdr = '';
|
||||
if ($masterCacheList[$n][0] == 1) { # directory
|
||||
&cache_speech("for folder $masterCacheList[$n][1]", $masterCacheList[$n][2]);
|
||||
} elsif ($masterCacheList[$n][0] == 2) { # file
|
||||
&cache_speech("to play $masterCacheList[$n][1]", $masterCacheList[$n][2]);
|
||||
} elsif ($masterCacheList[$n][0] == 3) { # number
|
||||
&cache_speech($masterCacheList[$n][1], $masterCacheList[$n][2]);
|
||||
}
|
||||
if (time() >= $theTime + 30) {
|
||||
my $percent = int($n / @masterCacheList * 100);
|
||||
print "SAY NUMBER $percent \"\"\n";
|
||||
my $result = <STDIN>; &check_result($result);
|
||||
print "EXEC Playback \"$FESTIVALCACHE/jukebox_percent\"\n";
|
||||
my $result = <STDIN>; &check_result($result);
|
||||
$theTime = time();
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
# this function will fill the @masterCacheList of all the files that
|
||||
# need to have text2speeced ulaw files of their names generated
|
||||
sub music_dir_cache_genlist {
|
||||
my $dir = shift;
|
||||
if (!opendir(THEDIR, rmts($MUSIC.$dir))) {
|
||||
print STDERR "Failed to open music directory: $dir\n";
|
||||
exit 1;
|
||||
}
|
||||
my @files = sort readdir THEDIR;
|
||||
my $foundFiles = 0;
|
||||
my $tmpMaxNum = 0;
|
||||
foreach my $file (@files) {
|
||||
chomp;
|
||||
if ($file ne '.' && $file ne '..' && $file ne 'festivalcache') { # ignore special files
|
||||
my $real_version = &rmts($MUSIC.$dir).'/'.$file;
|
||||
my $cache_version = &rmts($FESTIVALCACHE.$dir).'/'.$file.'.ul';
|
||||
my $cache_version2 = &rmts($FESTIVALCACHE.$dir).'/'.$file;
|
||||
my $cache_version_esc = &clean_file($cache_version);
|
||||
my $cache_version2_esc = &clean_file($cache_version2);
|
||||
|
||||
if (-d $real_version) {
|
||||
if (music_dir_cache_genlist(rmts($dir).'/'.$file)) {
|
||||
$tmpMaxNum++;
|
||||
$maxNumber = $tmpMaxNum if $tmpMaxNum > $maxNumber;
|
||||
push(@masterCacheList, [1, $file, $cache_version_esc]) if ! -e $cache_version_esc;
|
||||
$foundFiles = 1;
|
||||
}
|
||||
} elsif ($real_version =~ /\.mp3$/) {
|
||||
$tmpMaxNum++;
|
||||
$maxNumber = $tmpMaxNum if $tmpMaxNum > $maxNumber;
|
||||
push(@masterCacheList, [2, &remove_file_extension($file), $cache_version_esc]) if ! -e $cache_version_esc;
|
||||
$foundFiles = 1;
|
||||
}
|
||||
}
|
||||
}
|
||||
close(THEDIR);
|
||||
return $foundFiles;
|
||||
}
|
||||
|
||||
sub rmts { # remove trailing slash
|
||||
my $hog = shift;
|
||||
$hog =~ s/\/$//;
|
||||
return $hog;
|
||||
}
|
||||
|
||||
sub extract_file_name {
|
||||
my $hog = shift;
|
||||
$hog =~ /\/?([^\/]+)$/;
|
||||
return $1;
|
||||
}
|
||||
|
||||
sub extract_file_dir {
|
||||
my $hog = shift;
|
||||
return $hog if ! ($hog =~ /\//);
|
||||
$hog =~ /(.*)\/[^\/]*$/;
|
||||
return $1;
|
||||
}
|
||||
|
||||
sub remove_file_extension {
|
||||
my $hog = shift;
|
||||
return $hog if ! ($hog =~ /\./);
|
||||
$hog =~ /(.*)\.[^.]*$/;
|
||||
return $1;
|
||||
}
|
||||
|
||||
sub clean_file {
|
||||
my $hog = shift;
|
||||
$hog =~ s/\\/_/g;
|
||||
$hog =~ s/ /_/g;
|
||||
$hog =~ s/\t/_/g;
|
||||
$hog =~ s/\'/_/g;
|
||||
$hog =~ s/\"/_/g;
|
||||
$hog =~ s/\(/_/g;
|
||||
$hog =~ s/\)/_/g;
|
||||
$hog =~ s/&/_/g;
|
||||
$hog =~ s/\[/_/g;
|
||||
$hog =~ s/\]/_/g;
|
||||
$hog =~ s/\$/_/g;
|
||||
$hog =~ s/\|/_/g;
|
||||
$hog =~ s/\^/_/g;
|
||||
return $hog;
|
||||
}
|
||||
|
||||
sub remove_special_chars {
|
||||
my $hog = shift;
|
||||
$hog =~ s/\\//g;
|
||||
$hog =~ s/ //g;
|
||||
$hog =~ s/\t//g;
|
||||
$hog =~ s/\'//g;
|
||||
$hog =~ s/\"//g;
|
||||
$hog =~ s/\(//g;
|
||||
$hog =~ s/\)//g;
|
||||
$hog =~ s/&//g;
|
||||
$hog =~ s/\[//g;
|
||||
$hog =~ s/\]//g;
|
||||
$hog =~ s/\$//g;
|
||||
$hog =~ s/\|//g;
|
||||
$hog =~ s/\^//g;
|
||||
return $hog;
|
||||
}
|
||||
|
||||
sub escape_file {
|
||||
my $hog = shift;
|
||||
$hog =~ s/\\/\\\\/g;
|
||||
$hog =~ s/ /\\ /g;
|
||||
$hog =~ s/\t/\\\t/g;
|
||||
$hog =~ s/\'/\\\'/g;
|
||||
$hog =~ s/\"/\\\"/g;
|
||||
$hog =~ s/\(/\\\(/g;
|
||||
$hog =~ s/\)/\\\)/g;
|
||||
$hog =~ s/&/\\&/g;
|
||||
$hog =~ s/\[/\\\[/g;
|
||||
$hog =~ s/\]/\\\]/g;
|
||||
$hog =~ s/\$/\\\$/g;
|
||||
$hog =~ s/\|/\\\|/g;
|
||||
$hog =~ s/\^/\\\^/g;
|
||||
return $hog;
|
||||
}
|
||||
|
||||
sub check_result {
|
||||
my ($res) = @_;
|
||||
my $retval;
|
||||
$tests++;
|
||||
chomp $res;
|
||||
if ($res =~ /^200/) {
|
||||
$res =~ /result=(-?\d+)/;
|
||||
if (!length($1)) {
|
||||
print STDERR "FAIL ($res)\n";
|
||||
$fail++;
|
||||
exit 1;
|
||||
} else {
|
||||
print STDERR "PASS ($1)\n";
|
||||
return $1;
|
||||
}
|
||||
} else {
|
||||
print STDERR "FAIL (unexpected result '$res')\n";
|
||||
exit 1;
|
||||
}
|
||||
}
|
||||
@@ -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);
|
||||
@@ -1,33 +1,17 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
* Asterisk -- A telephony toolkit for Linux.
|
||||
*
|
||||
* Copyright (C) 1999 - 2005, Digium, Inc.
|
||||
* u-Law to Signed linear conversion
|
||||
*
|
||||
* Copyright (C) 1999, Mark Spencer
|
||||
*
|
||||
* Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
* Mark Spencer <markster@linux-support.net>
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief u-Law to Signed linear conversion
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include "asterisk/alaw.h"
|
||||
#include <asterisk/alaw.h>
|
||||
|
||||
#define AMI_MASK 0x55
|
||||
|
||||
43
app.c
Normal file
43
app.c
Normal file
@@ -0,0 +1,43 @@
|
||||
/*
|
||||
* Asterisk -- A telephony toolkit for Linux.
|
||||
*
|
||||
* Channel Management
|
||||
*
|
||||
* Copyright (C) 1999, Mark Spencer
|
||||
*
|
||||
* Mark Spencer <markster@linux-support.net>
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <pthread.h>
|
||||
#include <string.h>
|
||||
#include <sys/time.h>
|
||||
#include <signal.h>
|
||||
#include <errno.h>
|
||||
#include <unistd.h>
|
||||
#include <asterisk/channel.h>
|
||||
#include <asterisk/file.h>
|
||||
#include <asterisk/app.h>
|
||||
|
||||
/* set timeout to 0 for "standard" timeouts. Set timeout to -1 for
|
||||
"ludicrous time" (essentially never times out) */
|
||||
int ast_app_getdata(struct ast_channel *c, char *prompt, char *s, int maxlen, int timeout)
|
||||
{
|
||||
int res,to,fto;
|
||||
if (prompt) {
|
||||
res = ast_streamfile(c, prompt, c->language);
|
||||
if (res < 0)
|
||||
return res;
|
||||
}
|
||||
fto = 6000;
|
||||
to = 2000;
|
||||
if (timeout > 0) fto = to = timeout;
|
||||
if (timeout < 0) fto = to = 1000000000;
|
||||
res = ast_readstring(c, s, maxlen, to, fto, "#");
|
||||
return res;
|
||||
}
|
||||
|
||||
@@ -1,41 +1,42 @@
|
||||
#
|
||||
# Asterisk -- A telephony toolkit for Linux.
|
||||
#
|
||||
# Makefile for PBX applications
|
||||
# Makefile for PBX frontends (dynamically loaded)
|
||||
#
|
||||
# Copyright (C) 1999-2006, Digium, Inc.
|
||||
# Copyright (C) 1999, Mark Spencer
|
||||
#
|
||||
# Mark Spencer <markster@linux-support.net>
|
||||
#
|
||||
# This program is free software, distributed under the terms of
|
||||
# the GNU General Public License
|
||||
#
|
||||
|
||||
-include ../menuselect.makeopts ../menuselect.makedeps
|
||||
#APPS=app_dial.so app_playback.so app_directory.so app_intercom.so app_mp3.so
|
||||
APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_intercom.so app_mp3.so \
|
||||
app_system.so app_echo.so app_record.so app_image.so app_url.so app_disa.so \
|
||||
app_agi.so app_qcall.so
|
||||
|
||||
C_MODS:=$(filter-out $(MENUSELECT_APPS),$(patsubst %.c,%,$(wildcard app_*.c)))
|
||||
CC_MODS:=$(filter-out $(MENUSELECT_APPS),$(patsubst %.cc,%,$(wildcard app_*.cc)))
|
||||
APPS+=$(shell if [ -f /usr/include/zap.h ]; then echo "app_zapras.so" ; fi)
|
||||
|
||||
LOADABLE_MODS:=$(C_MODS) $(CC_MODS)
|
||||
CFLAGS+=
|
||||
|
||||
ifneq ($(findstring apps,$(MENUSELECT_EMBED)),)
|
||||
EMBEDDED_MODS:=$(LOADABLE_MODS)
|
||||
LOADABLE_MODS:=
|
||||
endif
|
||||
all: $(APPS)
|
||||
|
||||
MENUSELECT_OPTS_app_directory:=$(MENUSELECT_OPTS_app_voicemail)
|
||||
ifneq ($(findstring ODBC_STORAGE,$(MENUSELECT_OPTS_app_voicemail)),)
|
||||
MENUSELECT_DEPENDS_app_voicemail+=$(MENUSELECT_DEPENDS_ODBC_STORAGE)
|
||||
MENUSELECT_DEPENDS_app_directory+=$(MENUSELECT_DEPENDS_ODBC_STORAGE)
|
||||
endif
|
||||
ifneq ($(findstring IMAP_STORAGE,$(MENUSELECT_OPTS_app_voicemail)),)
|
||||
MENUSELECT_DEPENDS_app_voicemail+=$(MENUSELECT_DEPENDS_IMAP_STORAGE)
|
||||
MENUSELECT_DEPENDS_app_directory+=$(MENUSELECT_DEPENDS_IMAP_STORAGE)
|
||||
endif
|
||||
clean:
|
||||
rm -f *.so *.o look
|
||||
|
||||
ifeq (SunOS,$(shell uname))
|
||||
MENUSELECT_DEPENDS_app_chanspy+=RT
|
||||
RT_LIB=-lrt
|
||||
endif
|
||||
%.so : %.o
|
||||
$(CC) -shared -Xlinker -x -o $@ $<
|
||||
|
||||
all: _all
|
||||
install: all
|
||||
for x in $(APPS); do $(INSTALL) -m 755 $$x $(MODULES_DIR) ; done
|
||||
|
||||
include $(ASTTOPDIR)/Makefile.moddir_rules
|
||||
app_todd.o: app_todd.c
|
||||
gcc -pipe -O6 -g -Iinclude -I../include -D_REENTRANT -march=i586 -DDO_CRASH -DDEBUG_THREADS -c -o app_todd.o app_todd.c
|
||||
|
||||
app_todd.so: app_todd.o
|
||||
$(CC) -shared -Xlinker -x -o $@ $< -L/usr/local/ssl/lib -lssl -lcrypto
|
||||
|
||||
|
||||
look: look.c
|
||||
gcc -pipe -O6 -g look.c -o look -lncurses
|
||||
|
||||
1590
apps/app_adsiprog.c
1590
apps/app_adsiprog.c
File diff suppressed because it is too large
Load Diff
873
apps/app_agi.c
Normal file
873
apps/app_agi.c
Normal file
@@ -0,0 +1,873 @@
|
||||
/*
|
||||
* Asterisk -- A telephony toolkit for Linux.
|
||||
*
|
||||
* Asterisk Gateway Interface
|
||||
*
|
||||
* Copyright (C) 1999, Mark Spencer
|
||||
*
|
||||
* Mark Spencer <markster@linux-support.net>
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
#include <asterisk/file.h>
|
||||
#include <asterisk/logger.h>
|
||||
#include <asterisk/channel.h>
|
||||
#include <asterisk/pbx.h>
|
||||
#include <asterisk/module.h>
|
||||
#include <math.h>
|
||||
#include <stdlib.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
#include <stdlib.h>
|
||||
#include <sys/signal.h>
|
||||
#include <sys/time.h>
|
||||
#include <asterisk/cli.h>
|
||||
#include <asterisk/logger.h>
|
||||
#include <asterisk/options.h>
|
||||
#include <asterisk/image.h>
|
||||
#include <asterisk/say.h>
|
||||
#include "../asterisk.h"
|
||||
|
||||
#include <pthread.h>
|
||||
|
||||
#define MAX_ARGS 128
|
||||
|
||||
/* Recycle some stuff from the CLI interface */
|
||||
#define fdprintf ast_cli
|
||||
|
||||
typedef struct agi_command {
|
||||
/* Null terminated list of the words of the command */
|
||||
char *cmda[AST_MAX_CMD_LEN];
|
||||
/* Handler for the command (fd for output, # of arguments, argument list).
|
||||
Returns RESULT_SHOWUSAGE for improper arguments */
|
||||
int (*handler)(struct ast_channel *chan, int fd, int argc, char *argv[]);
|
||||
/* Summary of the command (< 60 characters) */
|
||||
char *summary;
|
||||
/* Detailed usage information */
|
||||
char *usage;
|
||||
} agi_command;
|
||||
|
||||
static char *tdesc = "Asterisk Gateway Interface (AGI)";
|
||||
|
||||
static char *app = "AGI";
|
||||
|
||||
static char *synopsis = "Executes an AGI compliant application";
|
||||
|
||||
static char *descrip =
|
||||
" AGI(command|args): Executes an Asterisk Gateway Interface compliant\n"
|
||||
"program on a channel. AGI allows Asterisk to launch external programs\n"
|
||||
"written in any language to control a telephony channel, play audio,\n"
|
||||
"read DTMF digits, etc. by communicating with the AGI protocol on stdin\n"
|
||||
"and stdout. Returns -1 on hangup or if application requested hangup, or\n"
|
||||
"0 on non-hangup exit.\n";
|
||||
|
||||
STANDARD_LOCAL_USER;
|
||||
|
||||
LOCAL_USER_DECL;
|
||||
|
||||
#define TONE_BLOCK_SIZE 200
|
||||
|
||||
static float loudness = 8192.0;
|
||||
|
||||
unsigned char linear2ulaw(short sample);
|
||||
static void make_tone_block(unsigned char *data, float f1, int *x);
|
||||
|
||||
static void make_tone_block(unsigned char *data, float f1, int *x)
|
||||
{
|
||||
int i;
|
||||
float val;
|
||||
|
||||
for(i = 0; i < TONE_BLOCK_SIZE; i++)
|
||||
{
|
||||
val = loudness * sin((f1 * 2.0 * M_PI * (*x)++)/8000.0);
|
||||
data[i] = linear2ulaw((int)val);
|
||||
}
|
||||
/* wrap back around from 8000 */
|
||||
if (*x >= 8000) *x = 0;
|
||||
return;
|
||||
}
|
||||
|
||||
static int launch_script(char *script, char *args, int *fds, int *opid)
|
||||
{
|
||||
char tmp[256];
|
||||
int pid;
|
||||
int toast[2];
|
||||
int fromast[2];
|
||||
int x;
|
||||
if (script[0] != '/') {
|
||||
snprintf(tmp, sizeof(tmp), "%s/%s", AST_AGI_DIR, script);
|
||||
script = tmp;
|
||||
}
|
||||
if (pipe(toast)) {
|
||||
ast_log(LOG_WARNING, "Unable to create toast pipe: %s\n",strerror(errno));
|
||||
return -1;
|
||||
}
|
||||
if (pipe(fromast)) {
|
||||
ast_log(LOG_WARNING, "unable to create fromast pipe: %s\n", strerror(errno));
|
||||
close(toast[0]);
|
||||
close(toast[1]);
|
||||
return -1;
|
||||
}
|
||||
pid = fork();
|
||||
if (pid < 0) {
|
||||
ast_log(LOG_WARNING, "Failed to fork(): %s\n", strerror(errno));
|
||||
return -1;
|
||||
}
|
||||
if (!pid) {
|
||||
/* Redirect stdin and out */
|
||||
dup2(fromast[0], STDIN_FILENO);
|
||||
dup2(toast[1], STDOUT_FILENO);
|
||||
/* Close everything but stdin/out/error */
|
||||
for (x=STDERR_FILENO + 1;x<1024;x++)
|
||||
close(x);
|
||||
/* Execute script */
|
||||
execl(script, script, args, NULL);
|
||||
ast_log(LOG_WARNING, "Failed to execute '%s': %s\n", script, strerror(errno));
|
||||
exit(1);
|
||||
}
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "Launched AGI Script %s\n", script);
|
||||
fds[0] = toast[0];
|
||||
fds[1] = fromast[1];
|
||||
/* close what we're not using in the parent */
|
||||
close(toast[1]);
|
||||
close(fromast[0]);
|
||||
*opid = pid;
|
||||
return 0;
|
||||
|
||||
}
|
||||
|
||||
static void setup_env(struct ast_channel *chan, char *request, int fd)
|
||||
{
|
||||
/* Print initial environment, with agi_request always being the first
|
||||
thing */
|
||||
fdprintf(fd, "agi_request: %s\n", request);
|
||||
fdprintf(fd, "agi_channel: %s\n", chan->name);
|
||||
fdprintf(fd, "agi_language: %s\n", chan->language);
|
||||
fdprintf(fd, "agi_type: %s\n", chan->type);
|
||||
|
||||
/* ANI/DNIS */
|
||||
fdprintf(fd, "agi_callerid: %s\n", chan->callerid ? chan->callerid : "");
|
||||
fdprintf(fd, "agi_dnid: %s\n", chan->dnid ? chan->dnid : "");
|
||||
|
||||
/* Context information */
|
||||
fdprintf(fd, "agi_context: %s\n", chan->context);
|
||||
fdprintf(fd, "agi_extension: %s\n", chan->exten);
|
||||
fdprintf(fd, "agi_priority: %d\n", chan->priority);
|
||||
|
||||
/* End with empty return */
|
||||
fdprintf(fd, "\n");
|
||||
}
|
||||
|
||||
static int handle_waitfordigit(struct ast_channel *chan, int fd, int argc, char *argv[])
|
||||
{
|
||||
int res;
|
||||
int to;
|
||||
if (argc != 4)
|
||||
return RESULT_SHOWUSAGE;
|
||||
if (sscanf(argv[3], "%i", &to) != 1)
|
||||
return RESULT_SHOWUSAGE;
|
||||
res = ast_waitfordigit(chan, to);
|
||||
fdprintf(fd, "200 result=%d\n", res);
|
||||
if (res >= 0)
|
||||
return RESULT_SUCCESS;
|
||||
else
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
|
||||
static int handle_sendtext(struct ast_channel *chan, int fd, int argc, char *argv[])
|
||||
{
|
||||
int res;
|
||||
if (argc != 3)
|
||||
return RESULT_SHOWUSAGE;
|
||||
/* At the moment, the parser (perhaps broken) returns with
|
||||
the last argument PLUS the newline at the end of the input
|
||||
buffer. This probably needs to be fixed, but I wont do that
|
||||
because other stuff may break as a result. The right way
|
||||
would probably be to strip off the trailing newline before
|
||||
parsing, then here, add a newline at the end of the string
|
||||
before sending it to ast_sendtext --DUDE */
|
||||
res = ast_sendtext(chan, argv[2]);
|
||||
fdprintf(fd, "200 result=%d\n", res);
|
||||
if (res >= 0)
|
||||
return RESULT_SUCCESS;
|
||||
else
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
|
||||
static int handle_recvchar(struct ast_channel *chan, int fd, int argc, char *argv[])
|
||||
{
|
||||
int res;
|
||||
if (argc != 3)
|
||||
return RESULT_SHOWUSAGE;
|
||||
res = ast_recvchar(chan,atoi(argv[2]));
|
||||
if (res == 0) {
|
||||
fdprintf(fd, "200 result=%d (timeout)\n", res);
|
||||
return RESULT_SUCCESS;
|
||||
}
|
||||
if (res > 0) {
|
||||
fdprintf(fd, "200 result=%d\n", res);
|
||||
return RESULT_SUCCESS;
|
||||
}
|
||||
else {
|
||||
fdprintf(fd, "200 result=%d (hangup)\n", res);
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
}
|
||||
|
||||
static int handle_tddmode(struct ast_channel *chan, int fd, int argc, char *argv[])
|
||||
{
|
||||
int res,x;
|
||||
if (argc != 3)
|
||||
return RESULT_SHOWUSAGE;
|
||||
if (!strncasecmp(argv[2],"on",2)) x = 1; else x = 0;
|
||||
res = ast_channel_setoption(chan,AST_OPTION_TDD,&x,sizeof(char),0);
|
||||
fdprintf(fd, "200 result=%d\n", res);
|
||||
if (res >= 0)
|
||||
return RESULT_SUCCESS;
|
||||
else
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
|
||||
static int handle_sendimage(struct ast_channel *chan, int fd, int argc, char *argv[])
|
||||
{
|
||||
int res;
|
||||
if (argc != 3)
|
||||
return RESULT_SHOWUSAGE;
|
||||
res = ast_send_image(chan, argv[2]);
|
||||
if (!ast_check_hangup(chan))
|
||||
res = 0;
|
||||
fdprintf(fd, "200 result=%d\n", res);
|
||||
if (res >= 0)
|
||||
return RESULT_SUCCESS;
|
||||
else
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
|
||||
static int handle_streamfile(struct ast_channel *chan, int fd, int argc, char *argv[])
|
||||
{
|
||||
int res;
|
||||
if (argc != 4)
|
||||
return RESULT_SHOWUSAGE;
|
||||
res = ast_streamfile(chan, argv[2],chan->language);
|
||||
if (res) {
|
||||
fdprintf(fd, "200 result=%d\n", res);
|
||||
if (res >= 0)
|
||||
return RESULT_SHOWUSAGE;
|
||||
else
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
res = ast_waitstream(chan, argv[3]);
|
||||
|
||||
fdprintf(fd, "200 result=%d\n", res);
|
||||
if (res >= 0)
|
||||
return RESULT_SUCCESS;
|
||||
else
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
|
||||
static int handle_saynumber(struct ast_channel *chan, int fd, int argc, char *argv[])
|
||||
{
|
||||
int res;
|
||||
int num;
|
||||
if (argc != 4)
|
||||
return RESULT_SHOWUSAGE;
|
||||
if (sscanf(argv[2], "%i", &num) != 1)
|
||||
return RESULT_SHOWUSAGE;
|
||||
res = ast_say_number(chan, num, AST_DIGIT_ANY, chan->language);
|
||||
fdprintf(fd, "200 result=%d\n", res);
|
||||
if (res >= 0)
|
||||
return RESULT_SUCCESS;
|
||||
else
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
|
||||
int ast_app_getdata(struct ast_channel *c, char *prompt, char *s, int maxlen, int timeout);
|
||||
|
||||
static int handle_getdata(struct ast_channel *chan, int fd, int argc, char *argv[])
|
||||
{
|
||||
int res;
|
||||
char data[50];
|
||||
int max;
|
||||
int timeout;
|
||||
|
||||
if (argc < 3)
|
||||
return RESULT_SHOWUSAGE;
|
||||
if (argc >= 4) timeout = atoi(argv[3]); else timeout = 0;
|
||||
if (argc >= 5) max = atoi(argv[4]); else max = 50;
|
||||
res = ast_app_getdata(chan, argv[2], data, max, timeout);
|
||||
if (res == 1)
|
||||
fdprintf(fd, "200 result=%s (timeout)\n", data);
|
||||
else
|
||||
fdprintf(fd, "200 result=%s\n", data);
|
||||
if (res >= 0)
|
||||
return RESULT_SUCCESS;
|
||||
else
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
|
||||
static int handle_setcontext(struct ast_channel *chan, int fd, int argc, char *argv[])
|
||||
{
|
||||
|
||||
if (argc != 3)
|
||||
return RESULT_SHOWUSAGE;
|
||||
strncpy(chan->context, argv[2], sizeof(chan->context)-1);
|
||||
fdprintf(fd, "200 result=0\n");
|
||||
return RESULT_SUCCESS;
|
||||
}
|
||||
|
||||
static int handle_setextension(struct ast_channel *chan, int fd, int argc, char **argv)
|
||||
{
|
||||
if (argc != 3)
|
||||
return RESULT_SHOWUSAGE;
|
||||
strncpy(chan->exten, argv[2], sizeof(chan->exten)-1);
|
||||
fdprintf(fd, "200 result=0\n");
|
||||
return RESULT_SUCCESS;
|
||||
}
|
||||
|
||||
static int handle_setpriority(struct ast_channel *chan, int fd, int argc, char **argv)
|
||||
{
|
||||
int pri;
|
||||
if (argc != 3)
|
||||
return RESULT_SHOWUSAGE;
|
||||
if (sscanf(argv[2], "%i", &pri) != 1)
|
||||
return RESULT_SHOWUSAGE;
|
||||
chan->priority = pri - 1;
|
||||
fdprintf(fd, "200 result=0\n");
|
||||
return RESULT_SUCCESS;
|
||||
}
|
||||
|
||||
static int ms_diff(struct timeval *tv1, struct timeval *tv2)
|
||||
{
|
||||
int ms;
|
||||
|
||||
ms = (tv1->tv_sec - tv2->tv_sec) * 1000;
|
||||
ms += (tv1->tv_usec - tv2->tv_usec) / 1000;
|
||||
return(ms);
|
||||
}
|
||||
|
||||
static int handle_recordfile(struct ast_channel *chan, int fd, int argc, char *argv[])
|
||||
{
|
||||
struct ast_filestream *fs;
|
||||
struct ast_frame *f,wf;
|
||||
struct timeval tv, start, lastout, now, notime = { 0,0 } ;
|
||||
fd_set readfds;
|
||||
unsigned char tone_block[TONE_BLOCK_SIZE];
|
||||
int res = -1;
|
||||
int ms,i,j;
|
||||
|
||||
if (argc < 6)
|
||||
return RESULT_SHOWUSAGE;
|
||||
if (sscanf(argv[5], "%i", &ms) != 1)
|
||||
return RESULT_SHOWUSAGE;
|
||||
|
||||
if (argc > 6) { /* if to beep */
|
||||
i = 0;
|
||||
lastout.tv_sec = lastout.tv_usec = 0;
|
||||
for(j = 0; j < 13; j++)
|
||||
{
|
||||
do gettimeofday(&now,NULL);
|
||||
while (lastout.tv_sec &&
|
||||
(ms_diff(&now,&lastout) < 25));
|
||||
lastout.tv_sec = now.tv_sec;
|
||||
lastout.tv_usec = now.tv_usec;
|
||||
wf.frametype = AST_FRAME_VOICE;
|
||||
wf.subclass = AST_FORMAT_ULAW;
|
||||
wf.offset = AST_FRIENDLY_OFFSET;
|
||||
wf.mallocd = 0;
|
||||
wf.data = tone_block;
|
||||
wf.datalen = TONE_BLOCK_SIZE;
|
||||
/* make this tone block */
|
||||
make_tone_block(tone_block,1000.0,&i);
|
||||
wf.timelen = wf.datalen / 8;
|
||||
if (ast_write(chan, &wf)) {
|
||||
fdprintf(fd, "200 result=%d (hangup)\n", 0);
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
FD_ZERO(&readfds);
|
||||
FD_SET(chan->fds[0],&readfds);
|
||||
/* if no read avail, do send again */
|
||||
if (select(chan->fds[0] + 1,&readfds,NULL,
|
||||
NULL,¬ime) < 1) continue;
|
||||
f = ast_read(chan);
|
||||
if (!f) {
|
||||
fdprintf(fd, "200 result=%d (hangup)\n", 0);
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
switch(f->frametype) {
|
||||
case AST_FRAME_DTMF:
|
||||
if (strchr(argv[4], f->subclass)) {
|
||||
/* This is an interrupting chracter */
|
||||
fdprintf(fd, "200 result=%d (dtmf)\n", f->subclass);
|
||||
ast_frfree(f);
|
||||
return RESULT_SUCCESS;
|
||||
}
|
||||
break;
|
||||
case AST_FRAME_VOICE:
|
||||
break; /* throw it away */
|
||||
}
|
||||
ast_frfree(f);
|
||||
}
|
||||
/* suck in 5 voice frames to make up for echo of beep, etc */
|
||||
for(i = 0; i < 5; i++) {
|
||||
f = ast_read(chan);
|
||||
if (!f) {
|
||||
fdprintf(fd, "200 result=%d (hangup)\n", 0);
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
switch(f->frametype) {
|
||||
case AST_FRAME_DTMF:
|
||||
if (strchr(argv[4], f->subclass)) {
|
||||
/* This is an interrupting chracter */
|
||||
fdprintf(fd, "200 result=%d (dtmf)\n", f->subclass);
|
||||
ast_frfree(f);
|
||||
return RESULT_SUCCESS;
|
||||
}
|
||||
break;
|
||||
case AST_FRAME_VOICE:
|
||||
break; /* throw it away */
|
||||
}
|
||||
ast_frfree(f);
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
fs = ast_writefile(argv[2], argv[3], NULL, O_CREAT | O_TRUNC | O_WRONLY, 0, 0644);
|
||||
if (!fs) {
|
||||
fdprintf(fd, "200 result=%d (writefile)\n", res);
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
gettimeofday(&start, NULL);
|
||||
gettimeofday(&tv, NULL);
|
||||
while ((ms < 0) || (((tv.tv_sec - start.tv_sec) * 1000 + (tv.tv_usec - start.tv_usec)/1000) < ms)) {
|
||||
res = ast_waitfor(chan, -1);
|
||||
if (res < 0) {
|
||||
ast_closestream(fs);
|
||||
fdprintf(fd, "200 result=%d (waitfor)\n", res);
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
f = ast_read(chan);
|
||||
if (!f) {
|
||||
fdprintf(fd, "200 result=%d (hangup)\n", 0);
|
||||
ast_closestream(fs);
|
||||
return RESULT_FAILURE;
|
||||
}
|
||||
switch(f->frametype) {
|
||||
case AST_FRAME_DTMF:
|
||||
if (strchr(argv[4], f->subclass)) {
|
||||
/* This is an interrupting chracter */
|
||||
fdprintf(fd, "200 result=%d (dtmf)\n", f->subclass);
|
||||
ast_closestream(fs);
|
||||
ast_frfree(f);
|
||||
return RESULT_SUCCESS;
|
||||
}
|
||||
break;
|
||||
case AST_FRAME_VOICE:
|
||||
ast_writestream(fs, f);
|
||||
break;
|
||||
};
|
||||
ast_frfree(f);
|
||||
gettimeofday(&tv, NULL);
|
||||
}
|
||||
fdprintf(fd, "200 result=%d (timeout)\n", 0);
|
||||
ast_closestream(fs);
|
||||
return RESULT_SUCCESS;
|
||||
}
|
||||
|
||||
static char usage_waitfordigit[] =
|
||||
" Usage: WAIT FOR DIGIT <timeout>\n"
|
||||
" Waits up to 'timeout' seconds for channel to receive a DTMF digit.\n"
|
||||
" Returns -1 on channel failure, 0 if no digit is received in the timeout, or\n"
|
||||
" the numerical value of the ascii of the digit if one is received. Use -1\n"
|
||||
" for the timeout value if you desire the call to block indefinitely.\n";
|
||||
|
||||
static char usage_sendtext[] =
|
||||
" Usage: SEND TEXT \"<text to send>\"\n"
|
||||
" Sends the given text on a channel. Most channels do not support the\n"
|
||||
" transmission of text. Returns 0 if text is sent, or if the channel does not\n"
|
||||
" support text transmission. Returns -1 only on error/hangup. Text\n"
|
||||
" consisting of greater than one word should be placed in quotes since the\n"
|
||||
" command only accepts a single argument.\n";
|
||||
|
||||
static char usage_recvchar[] =
|
||||
" Usage: RECEIVE CHAR <timeout>\n"
|
||||
" Receives a character of text on a channel. Specify timeout to be the\n"
|
||||
" maximum time to wait for input in milliseconds, or 0 for infinite. Most channels\n"
|
||||
" do not support the reception of text. Returns the decimal value of the character\n"
|
||||
" if one is received, or 0 if the channel does not support text reception. Returns\n"
|
||||
" -1 only on error/hangup.\n";
|
||||
|
||||
static char usage_tddmode[] =
|
||||
" Usage: TDD MODE <on|off>\n"
|
||||
" Enable/Disable TDD transmission/reception on a channel. Returns 1 if\n"
|
||||
" successful, or 0 if channel is not TDD-capable.\n";
|
||||
|
||||
static char usage_sendimage[] =
|
||||
" Usage: SEND IMAGE <image>\n"
|
||||
" Sends the given image on a channel. Most channels do not support the\n"
|
||||
" transmission of images. Returns 0 if image is sent, or if the channel does not\n"
|
||||
" support image transmission. Returns -1 only on error/hangup. Image names\n"
|
||||
" should not include extensions.\n";
|
||||
|
||||
static char usage_streamfile[] =
|
||||
" Usage: STREAM FILE <filename> <escape digits>\n"
|
||||
" Send the given file, allowing playback to be interrupted by the given\n"
|
||||
" digits, if any. Use double quotes for the digits if you wish none to be\n"
|
||||
" permitted. Returns 0 if playback completes without a digit being pressed, or\n"
|
||||
" the ASCII numerical value of the digit if one was pressed, or -1 on error or\n"
|
||||
" if the channel was disconnected. Remember, the file extension must not be\n"
|
||||
" included in the filename.\n";
|
||||
|
||||
static char usage_saynumber[] =
|
||||
" Usage: SAY NUMBER <number> <escape digits>\n"
|
||||
" Say a given number, returning early if any of the given DTMF digits\n"
|
||||
" are received on the channel. Returns 0 if playback completes without a digit\n"
|
||||
" being pressed, or the ASCII numerical value of the digit if one was pressed or\n"
|
||||
" -1 on error/hangup.\n";
|
||||
|
||||
static char usage_getdata[] =
|
||||
" Usage: GET DATA <file to be streamed> [timeout] [max digits]\n"
|
||||
" Stream the given file, and recieve DTMF data. Returns the digits recieved\n"
|
||||
"from the channel at the other end.\n";
|
||||
|
||||
static char usage_setcontext[] =
|
||||
" Usage: SET CONTEXT <desired context>\n"
|
||||
" Sets the context for continuation upon exiting the application.\n";
|
||||
|
||||
static char usage_setextension[] =
|
||||
" Usage: SET EXTENSION <new extension>\n"
|
||||
" Changes the extension for continuation upon exiting the application.\n";
|
||||
|
||||
static char usage_setpriority[] =
|
||||
" Usage: SET PRIORITY <num>\n"
|
||||
" Changes the priority for continuation upon exiting the application.\n";
|
||||
|
||||
static char usage_recordfile[] =
|
||||
" Usage: RECORD FILE <filename> <format> <escape digits> <timeout> [BEEP]\n"
|
||||
" Record to a file until a given dtmf digit in the sequence is received\n"
|
||||
" Returns -1 on hangup or error. The format will specify what kind of file\n"
|
||||
" will be recorded. The timeout is the maximum record time in milliseconds, or\n"
|
||||
" -1 for no timeout\n";
|
||||
|
||||
agi_command commands[] = {
|
||||
{ { "wait", "for", "digit", NULL }, handle_waitfordigit, "Waits for a digit to be pressed", usage_waitfordigit },
|
||||
{ { "send", "text", NULL }, handle_sendtext, "Sends text to channels supporting it", usage_sendtext },
|
||||
{ { "receive", "char", NULL }, handle_recvchar, "Receives text from channels supporting it", usage_recvchar },
|
||||
{ { "tdd", "mode", NULL }, handle_tddmode, "Sends text to channels supporting it", usage_tddmode },
|
||||
{ { "stream", "file", NULL }, handle_streamfile, "Sends audio file on channel", usage_streamfile },
|
||||
{ { "send", "image", NULL }, handle_sendimage, "Sends images to channels supporting it", usage_sendimage },
|
||||
{ { "say", "number", NULL }, handle_saynumber, "Says a given number", usage_saynumber },
|
||||
{ { "get", "data", NULL }, handle_getdata, "Gets data on a channel", usage_getdata },
|
||||
{ { "set", "context", NULL }, handle_setcontext, "Sets channel context", usage_setcontext },
|
||||
{ { "set", "extension", NULL }, handle_setextension, "Changes channel extension", usage_setextension },
|
||||
{ { "set", "priority", NULL }, handle_setpriority, "Prioritizes the channel", usage_setpriority },
|
||||
{ { "record", "file", NULL }, handle_recordfile, "Records to a given file", usage_recordfile }
|
||||
};
|
||||
|
||||
static agi_command *find_command(char *cmds[])
|
||||
{
|
||||
int x;
|
||||
int y;
|
||||
int match;
|
||||
for (x=0;x < sizeof(commands) / sizeof(commands[0]);x++) {
|
||||
/* start optimistic */
|
||||
match = 1;
|
||||
for (y=0;match && cmds[y]; y++) {
|
||||
/* If there are no more words in the command (and we're looking for
|
||||
an exact match) or there is a difference between the two words,
|
||||
then this is not a match */
|
||||
if (!commands[x].cmda[y])
|
||||
break;
|
||||
if (strcasecmp(commands[x].cmda[y], cmds[y]))
|
||||
match = 0;
|
||||
}
|
||||
/* If more words are needed to complete the command then this is not
|
||||
a candidate (unless we're looking for a really inexact answer */
|
||||
if (commands[x].cmda[y])
|
||||
match = 0;
|
||||
if (match)
|
||||
return &commands[x];
|
||||
}
|
||||
return NULL;
|
||||
}
|
||||
|
||||
|
||||
static int parse_args(char *s, int *max, char *argv[])
|
||||
{
|
||||
int x=0;
|
||||
int quoted=0;
|
||||
int escaped=0;
|
||||
int whitespace=1;
|
||||
char *cur;
|
||||
|
||||
cur = s;
|
||||
while(*s) {
|
||||
switch(*s) {
|
||||
case '"':
|
||||
/* If it's escaped, put a literal quote */
|
||||
if (escaped)
|
||||
goto normal;
|
||||
else
|
||||
quoted = !quoted;
|
||||
if (quoted && whitespace) {
|
||||
/* If we're starting a quote, coming off white space start a new word, too */
|
||||
argv[x++] = cur;
|
||||
whitespace=0;
|
||||
}
|
||||
escaped = 0;
|
||||
break;
|
||||
case ' ':
|
||||
case '\t':
|
||||
if (!quoted && !escaped) {
|
||||
/* If we're not quoted, mark this as whitespace, and
|
||||
end the previous argument */
|
||||
whitespace = 1;
|
||||
*(cur++) = '\0';
|
||||
} else
|
||||
/* Otherwise, just treat it as anything else */
|
||||
goto normal;
|
||||
break;
|
||||
case '\\':
|
||||
/* If we're escaped, print a literal, otherwise enable escaping */
|
||||
if (escaped) {
|
||||
goto normal;
|
||||
} else {
|
||||
escaped=1;
|
||||
}
|
||||
break;
|
||||
default:
|
||||
normal:
|
||||
if (whitespace) {
|
||||
if (x >= MAX_ARGS -1) {
|
||||
ast_log(LOG_WARNING, "Too many arguments, truncating\n");
|
||||
break;
|
||||
}
|
||||
/* Coming off of whitespace, start the next argument */
|
||||
argv[x++] = cur;
|
||||
whitespace=0;
|
||||
}
|
||||
*(cur++) = *s;
|
||||
escaped=0;
|
||||
}
|
||||
s++;
|
||||
}
|
||||
/* Null terminate */
|
||||
*(cur++) = '\0';
|
||||
argv[x] = NULL;
|
||||
*max = x;
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int agi_handle_command(struct ast_channel *chan, int fd, char *buf)
|
||||
{
|
||||
char *argv[MAX_ARGS];
|
||||
int argc = 0;
|
||||
int res;
|
||||
agi_command *c;
|
||||
argc = MAX_ARGS;
|
||||
parse_args(buf, &argc, argv);
|
||||
#if 0
|
||||
{ int x;
|
||||
for (x=0;x<argc;x++)
|
||||
fprintf(stderr, "Got Arg%d: %s\n", x, argv[x]); }
|
||||
#endif
|
||||
c = find_command(argv);
|
||||
if (c) {
|
||||
res = c->handler(chan, fd, argc, argv);
|
||||
switch(res) {
|
||||
case RESULT_SHOWUSAGE:
|
||||
fdprintf(fd, "520-Invalid command syntax. Proper usage follows:\n");
|
||||
fdprintf(fd, c->usage);
|
||||
fdprintf(fd, "520 End of proper usage.\n");
|
||||
break;
|
||||
case RESULT_FAILURE:
|
||||
/* They've already given the failure. We've been hung up on so handle this
|
||||
appropriately */
|
||||
return -1;
|
||||
}
|
||||
} else {
|
||||
fdprintf(fd, "510 Invalid or unknown command\n");
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int run_agi(struct ast_channel *chan, char *request, int *fds, int pid)
|
||||
{
|
||||
struct ast_channel *c;
|
||||
int outfd;
|
||||
int ms;
|
||||
int returnstatus = 0;
|
||||
struct ast_frame *f;
|
||||
char buf[2048];
|
||||
FILE *readf;
|
||||
if (!(readf = fdopen(fds[0], "r"))) {
|
||||
ast_log(LOG_WARNING, "Unable to fdopen file descriptor\n");
|
||||
kill(pid, SIGHUP);
|
||||
return -1;
|
||||
}
|
||||
setlinebuf(readf);
|
||||
setup_env(chan, request, fds[1]);
|
||||
for (;;) {
|
||||
ms = -1;
|
||||
c = ast_waitfor_nandfds(&chan, 1, &fds[0], 1, NULL, &outfd, &ms);
|
||||
if (c) {
|
||||
/* Idle the channel until we get a command */
|
||||
f = ast_read(c);
|
||||
if (!f) {
|
||||
ast_log(LOG_DEBUG, "%s hungup\n", chan->name);
|
||||
returnstatus = -1;
|
||||
break;
|
||||
} else {
|
||||
ast_frfree(f);
|
||||
}
|
||||
} else if (outfd > -1) {
|
||||
if (!fgets(buf, sizeof(buf), readf)) {
|
||||
/* Program terminated */
|
||||
if (returnstatus)
|
||||
returnstatus = -1;
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "AGI Script %s completed, returning %d\n", request, returnstatus);
|
||||
/* No need to kill the pid anymore, since they closed us */
|
||||
pid = -1;
|
||||
break;
|
||||
}
|
||||
#if 0
|
||||
/* Un-comment this code to fix the problem with
|
||||
the newline being included in the parsed
|
||||
command string(s) output --DUDE */
|
||||
/* get rid of trailing newline, if any */
|
||||
if (*buf && buf[strlen(buf) - 1] == '\n')
|
||||
buf[strlen(buf) - 1] = 0;
|
||||
#endif
|
||||
returnstatus |= agi_handle_command(chan, fds[1], buf);
|
||||
/* If the handle_command returns -1, we need to stop */
|
||||
if (returnstatus < 0) {
|
||||
break;
|
||||
}
|
||||
} else {
|
||||
ast_log(LOG_WARNING, "No channel, no fd?\n");
|
||||
returnstatus = -1;
|
||||
break;
|
||||
}
|
||||
}
|
||||
/* Notify process */
|
||||
if (pid > -1)
|
||||
kill(pid, SIGHUP);
|
||||
fclose(readf);
|
||||
return returnstatus;
|
||||
}
|
||||
|
||||
static int agi_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res=0;
|
||||
struct localuser *u;
|
||||
char *args,*ringy;
|
||||
char tmp[256];
|
||||
int fds[2];
|
||||
int pid;
|
||||
if (!data || !strlen(data)) {
|
||||
ast_log(LOG_WARNING, "AGI requires an argument (script)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
|
||||
strncpy(tmp, data, sizeof(tmp)-1);
|
||||
strtok(tmp, "|");
|
||||
args = strtok(NULL, "|");
|
||||
ringy = strtok(NULL,"|");
|
||||
if (!args)
|
||||
args = "";
|
||||
LOCAL_USER_ADD(u);
|
||||
/* Answer if need be */
|
||||
if (chan->state != AST_STATE_UP) {
|
||||
if (ringy) { /* if for ringing first */
|
||||
/* a little ringy-dingy first */
|
||||
ast_indicate(chan, AST_CONTROL_RINGING);
|
||||
sleep(3);
|
||||
}
|
||||
if (ast_answer(chan)) {
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
res = launch_script(tmp, args, fds, &pid);
|
||||
if (!res) {
|
||||
res = run_agi(chan, tmp, fds, pid);
|
||||
close(fds[0]);
|
||||
close(fds[1]);
|
||||
}
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
int unload_module(void)
|
||||
{
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, agi_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
|
||||
char *key()
|
||||
{
|
||||
return ASTERISK_GPL_KEY;
|
||||
}
|
||||
|
||||
#define CLIP 32635
|
||||
#define BIAS 0x84
|
||||
|
||||
unsigned char
|
||||
linear2ulaw(sample)
|
||||
short sample; {
|
||||
static int exp_lut[256] = {0,0,1,1,2,2,2,2,3,3,3,3,3,3,3,3,
|
||||
4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,
|
||||
5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
|
||||
5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
|
||||
6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
|
||||
6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
|
||||
6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
|
||||
6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
|
||||
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
|
||||
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
|
||||
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
|
||||
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
|
||||
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
|
||||
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
|
||||
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
|
||||
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7};
|
||||
int sign, exponent, mantissa;
|
||||
unsigned char ulawbyte;
|
||||
|
||||
/* Get the sample into sign-magnitude. */
|
||||
sign = (sample >> 8) & 0x80; /* set aside the sign */
|
||||
if (sign != 0) sample = -sample; /* get magnitude */
|
||||
if (sample > CLIP) sample = CLIP; /* clip the magnitude */
|
||||
|
||||
/* Convert from 16 bit linear to ulaw. */
|
||||
sample = sample + BIAS;
|
||||
exponent = exp_lut[(sample >> 7) & 0xFF];
|
||||
mantissa = (sample >> (exponent + 3)) & 0x0F;
|
||||
ulawbyte = ~(sign | (exponent << 4) | mantissa);
|
||||
#ifdef ZEROTRAP
|
||||
if (ulawbyte == 0) ulawbyte = 0x02; /* optional CCITT trap */
|
||||
#endif
|
||||
|
||||
return(ulawbyte);
|
||||
}
|
||||
@@ -1,841 +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. *
|
||||
*
|
||||
* *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING ***
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#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/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/translate.h"
|
||||
#include "asterisk/ulaw.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/dsp.h"
|
||||
#include "asterisk/config.h"
|
||||
#include "asterisk/localtime.h"
|
||||
#include "asterisk/callerid.h"
|
||||
#include "asterisk/astdb.h"
|
||||
#include "asterisk/utils.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 *app = "AlarmReceiver";
|
||||
|
||||
static char *synopsis = "Provide support for receiving 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";
|
||||
|
||||
/*
|
||||
* 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, "%u", &v);
|
||||
v++;
|
||||
|
||||
if(option_verbose >= 4)
|
||||
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: New value for %s: %u\n", key, v);
|
||||
|
||||
snprintf(value, sizeof(value), "%u", v);
|
||||
|
||||
res = ast_db_put(db_family, key, value);
|
||||
|
||||
if((res)&&(option_verbose >= 4))
|
||||
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: database_increment write error\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 = ast_calloc(1, sizeof(*enew)))) {
|
||||
res = -1;
|
||||
break;
|
||||
}
|
||||
|
||||
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 ast_module_user *u;
|
||||
event_node_t *elp, *efree;
|
||||
char signalling_type[64] = "";
|
||||
|
||||
event_node_t *event_head = NULL;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
/* 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);
|
||||
ast_module_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);
|
||||
ast_module_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) {
|
||||
ast_module_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);
|
||||
}
|
||||
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
/*
|
||||
* Load the configuration from the configuration file
|
||||
*/
|
||||
|
||||
static int load_config(void)
|
||||
{
|
||||
struct ast_config *cfg;
|
||||
const 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");
|
||||
return 0;
|
||||
}
|
||||
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 1;
|
||||
|
||||
}
|
||||
|
||||
/*
|
||||
* These functions are required to implement an Asterisk App.
|
||||
*/
|
||||
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
if(load_config())
|
||||
return ast_register_application(app, alarmreceiver_exec, synopsis, descrip);
|
||||
else
|
||||
return AST_MODULE_LOAD_DECLINE;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Alarm Receiver for Asterisk");
|
||||
398
apps/app_amd.c
398
apps/app_amd.c
@@ -1,398 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 2003 - 2006, Aheeva Technology.
|
||||
*
|
||||
* Claude Klimos (claude.klimos@aheeva.com)
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*
|
||||
* A license has been granted to Digium (via disclaimer) for the use of
|
||||
* this code.
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/dsp.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/config.h"
|
||||
#include "asterisk/app.h"
|
||||
|
||||
|
||||
static char *app = "AMD";
|
||||
static char *synopsis = "Attempts to detect answering machines";
|
||||
static char *descrip =
|
||||
" AMD([initialSilence][|greeting][|afterGreetingSilence][|totalAnalysisTime]\n"
|
||||
" [|minimumWordLength][|betweenWordsSilence][|maximumNumberOfWords]\n"
|
||||
" [|silenceThreshold])\n"
|
||||
" This application attempts to detect answering machines at the beginning\n"
|
||||
" of outbound calls. Simply call this application after the call\n"
|
||||
" has been answered (outbound only, of course).\n"
|
||||
" When loaded, AMD reads amd.conf and uses the parameters specified as\n"
|
||||
" default values. Those default values get overwritten when calling AMD\n"
|
||||
" with parameters.\n"
|
||||
"- 'initialSilence' is the maximum silence duration before the greeting. If\n"
|
||||
" exceeded then MACHINE.\n"
|
||||
"- 'greeting' is the maximum length of a greeting. If exceeded then MACHINE.\n"
|
||||
"- 'afterGreetingSilence' is the silence after detecting a greeting.\n"
|
||||
" If exceeded then HUMAN.\n"
|
||||
"- 'totalAnalysisTime' is the maximum time allowed for the algorithm to decide\n"
|
||||
" on a HUMAN or MACHINE.\n"
|
||||
"- 'minimumWordLength'is the minimum duration of Voice to considered as a word.\n"
|
||||
"- 'betweenWordsSilence' is the minimum duration of silence after a word to \n"
|
||||
" consider the audio that follows as a new word.\n"
|
||||
"- 'maximumNumberOfWords'is the maximum number of words in the greeting. \n"
|
||||
" If exceeded then MACHINE.\n"
|
||||
"- 'silenceThreshold' is the silence threshold.\n"
|
||||
"This application sets the following channel variable upon completion:\n"
|
||||
" AMDSTATUS - This is the status of the answering machine detection.\n"
|
||||
" Possible values are:\n"
|
||||
" MACHINE | HUMAN | NOTSURE | HANGUP\n"
|
||||
" AMDCAUSE - Indicates the cause that led to the conclusion.\n"
|
||||
" Possible values are:\n"
|
||||
" TOOLONG-<%d total_time>\n"
|
||||
" INITIALSILENCE-<%d silenceDuration>-<%d initialSilence>\n"
|
||||
" HUMAN-<%d silenceDuration>-<%d afterGreetingSilence>\n"
|
||||
" MAXWORDS-<%d wordsCount>-<%d maximumNumberOfWords>\n"
|
||||
" LONGGREETING-<%d voiceDuration>-<%d greeting>\n";
|
||||
|
||||
#define STATE_IN_WORD 1
|
||||
#define STATE_IN_SILENCE 2
|
||||
|
||||
/* Some default values for the algorithm parameters. These defaults will be overwritten from amd.conf */
|
||||
static int dfltInitialSilence = 2500;
|
||||
static int dfltGreeting = 1500;
|
||||
static int dfltAfterGreetingSilence = 800;
|
||||
static int dfltTotalAnalysisTime = 5000;
|
||||
static int dfltMinimumWordLength = 100;
|
||||
static int dfltBetweenWordsSilence = 50;
|
||||
static int dfltMaximumNumberOfWords = 3;
|
||||
static int dfltSilenceThreshold = 256;
|
||||
|
||||
static void isAnsweringMachine(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_frame *f = NULL;
|
||||
struct ast_dsp *silenceDetector = NULL;
|
||||
int dspsilence = 0, readFormat, framelength;
|
||||
int inInitialSilence = 1;
|
||||
int inGreeting = 0;
|
||||
int voiceDuration = 0;
|
||||
int silenceDuration = 0;
|
||||
int iTotalTime = 0;
|
||||
int iWordsCount = 0;
|
||||
int currentState = STATE_IN_SILENCE;
|
||||
int previousState = STATE_IN_SILENCE;
|
||||
int consecutiveVoiceDuration = 0;
|
||||
char amdCause[256] = "", amdStatus[256] = "";
|
||||
char *parse = ast_strdupa(data);
|
||||
|
||||
/* Lets set the initial values of the variables that will control the algorithm.
|
||||
The initial values are the default ones. If they are passed as arguments
|
||||
when invoking the application, then the default values will be overwritten
|
||||
by the ones passed as parameters. */
|
||||
int initialSilence = dfltInitialSilence;
|
||||
int greeting = dfltGreeting;
|
||||
int afterGreetingSilence = dfltAfterGreetingSilence;
|
||||
int totalAnalysisTime = dfltTotalAnalysisTime;
|
||||
int minimumWordLength = dfltMinimumWordLength;
|
||||
int betweenWordsSilence = dfltBetweenWordsSilence;
|
||||
int maximumNumberOfWords = dfltMaximumNumberOfWords;
|
||||
int silenceThreshold = dfltSilenceThreshold;
|
||||
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(argInitialSilence);
|
||||
AST_APP_ARG(argGreeting);
|
||||
AST_APP_ARG(argAfterGreetingSilence);
|
||||
AST_APP_ARG(argTotalAnalysisTime);
|
||||
AST_APP_ARG(argMinimumWordLength);
|
||||
AST_APP_ARG(argBetweenWordsSilence);
|
||||
AST_APP_ARG(argMaximumNumberOfWords);
|
||||
AST_APP_ARG(argSilenceThreshold);
|
||||
);
|
||||
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "AMD: %s %s %s (Fmt: %d)\n", chan->name ,chan->cid.cid_ani, chan->cid.cid_rdnis, chan->readformat);
|
||||
|
||||
/* Lets parse the arguments. */
|
||||
if (!ast_strlen_zero(parse)) {
|
||||
/* Some arguments have been passed. Lets parse them and overwrite the defaults. */
|
||||
AST_STANDARD_APP_ARGS(args, parse);
|
||||
if (!ast_strlen_zero(args.argInitialSilence))
|
||||
initialSilence = atoi(args.argInitialSilence);
|
||||
if (!ast_strlen_zero(args.argGreeting))
|
||||
greeting = atoi(args.argGreeting);
|
||||
if (!ast_strlen_zero(args.argAfterGreetingSilence))
|
||||
afterGreetingSilence = atoi(args.argAfterGreetingSilence);
|
||||
if (!ast_strlen_zero(args.argTotalAnalysisTime))
|
||||
totalAnalysisTime = atoi(args.argTotalAnalysisTime);
|
||||
if (!ast_strlen_zero(args.argMinimumWordLength))
|
||||
minimumWordLength = atoi(args.argMinimumWordLength);
|
||||
if (!ast_strlen_zero(args.argBetweenWordsSilence))
|
||||
betweenWordsSilence = atoi(args.argBetweenWordsSilence);
|
||||
if (!ast_strlen_zero(args.argMaximumNumberOfWords))
|
||||
maximumNumberOfWords = atoi(args.argMaximumNumberOfWords);
|
||||
if (!ast_strlen_zero(args.argSilenceThreshold))
|
||||
silenceThreshold = atoi(args.argSilenceThreshold);
|
||||
} else if (option_debug)
|
||||
ast_log(LOG_DEBUG, "AMD using the default parameters.\n");
|
||||
|
||||
/* Now we're ready to roll! */
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "AMD: initialSilence [%d] greeting [%d] afterGreetingSilence [%d] "
|
||||
"totalAnalysisTime [%d] minimumWordLength [%d] betweenWordsSilence [%d] maximumNumberOfWords [%d] silenceThreshold [%d] \n",
|
||||
initialSilence, greeting, afterGreetingSilence, totalAnalysisTime,
|
||||
minimumWordLength, betweenWordsSilence, maximumNumberOfWords, silenceThreshold );
|
||||
|
||||
/* Set read format to signed linear so we get signed linear frames in */
|
||||
readFormat = chan->readformat;
|
||||
if (ast_set_read_format(chan, AST_FORMAT_SLINEAR) < 0 ) {
|
||||
ast_log(LOG_WARNING, "AMD: Channel [%s]. Unable to set to linear mode, giving up\n", chan->name );
|
||||
pbx_builtin_setvar_helper(chan , "AMDSTATUS", "");
|
||||
pbx_builtin_setvar_helper(chan , "AMDCAUSE", "");
|
||||
return;
|
||||
}
|
||||
|
||||
/* Create a new DSP that will detect the silence */
|
||||
if (!(silenceDetector = ast_dsp_new())) {
|
||||
ast_log(LOG_WARNING, "AMD: Channel [%s]. Unable to create silence detector :(\n", chan->name );
|
||||
pbx_builtin_setvar_helper(chan , "AMDSTATUS", "");
|
||||
pbx_builtin_setvar_helper(chan , "AMDCAUSE", "");
|
||||
return;
|
||||
}
|
||||
|
||||
/* Set silence threshold to specified value */
|
||||
ast_dsp_set_threshold(silenceDetector, silenceThreshold);
|
||||
|
||||
/* Now we go into a loop waiting for frames from the channel */
|
||||
while ((res = ast_waitfor(chan, totalAnalysisTime)) > -1) {
|
||||
/* If we fail to read in a frame, that means they hung up */
|
||||
if (!(f = ast_read(chan))) {
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "AMD: HANGUP\n");
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "Got hangup\n");
|
||||
strcpy(amdStatus, "HANGUP");
|
||||
break;
|
||||
}
|
||||
|
||||
if (f->frametype == AST_FRAME_VOICE) {
|
||||
/* If the total time exceeds the analysis time then give up as we are not too sure */
|
||||
framelength = (ast_codec_get_samples(f) / DEFAULT_SAMPLES_PER_MS);
|
||||
iTotalTime += framelength;
|
||||
if (iTotalTime >= totalAnalysisTime) {
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "AMD: Channel [%s]. Too long...\n", chan->name );
|
||||
ast_frfree(f);
|
||||
strcpy(amdStatus , "NOTSURE");
|
||||
sprintf(amdCause , "TOOLONG-%d", iTotalTime);
|
||||
break;
|
||||
}
|
||||
|
||||
/* Feed the frame of audio into the silence detector and see if we get a result */
|
||||
dspsilence = 0;
|
||||
ast_dsp_silence(silenceDetector, f, &dspsilence);
|
||||
if (dspsilence) {
|
||||
silenceDuration = dspsilence;
|
||||
|
||||
if (silenceDuration >= betweenWordsSilence) {
|
||||
if (currentState != STATE_IN_SILENCE ) {
|
||||
previousState = currentState;
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "AMD: Changed state to STATE_IN_SILENCE\n");
|
||||
}
|
||||
currentState = STATE_IN_SILENCE;
|
||||
consecutiveVoiceDuration = 0;
|
||||
}
|
||||
|
||||
if (inInitialSilence == 1 && silenceDuration >= initialSilence) {
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "AMD: ANSWERING MACHINE: silenceDuration:%d initialSilence:%d\n",
|
||||
silenceDuration, initialSilence);
|
||||
ast_frfree(f);
|
||||
strcpy(amdStatus , "MACHINE");
|
||||
sprintf(amdCause , "INITIALSILENCE-%d-%d", silenceDuration, initialSilence);
|
||||
break;
|
||||
}
|
||||
|
||||
if (silenceDuration >= afterGreetingSilence && inGreeting == 1) {
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "AMD: HUMAN: silenceDuration:%d afterGreetingSilence:%d\n",
|
||||
silenceDuration, afterGreetingSilence);
|
||||
ast_frfree(f);
|
||||
strcpy(amdStatus , "HUMAN");
|
||||
sprintf(amdCause , "HUMAN-%d-%d", silenceDuration, afterGreetingSilence);
|
||||
break;
|
||||
}
|
||||
|
||||
} else {
|
||||
consecutiveVoiceDuration += framelength;
|
||||
voiceDuration += framelength;
|
||||
|
||||
/* If I have enough consecutive voice to say that I am in a Word, I can only increment the
|
||||
number of words if my previous state was Silence, which means that I moved into a word. */
|
||||
if (consecutiveVoiceDuration >= minimumWordLength && currentState == STATE_IN_SILENCE) {
|
||||
iWordsCount++;
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "AMD: Word detected. iWordsCount:%d\n", iWordsCount);
|
||||
previousState = currentState;
|
||||
currentState = STATE_IN_WORD;
|
||||
}
|
||||
|
||||
if (iWordsCount >= maximumNumberOfWords) {
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "AMD: ANSWERING MACHINE: iWordsCount:%d\n", iWordsCount);
|
||||
ast_frfree(f);
|
||||
strcpy(amdStatus , "MACHINE");
|
||||
sprintf(amdCause , "MAXWORDS-%d-%d", iWordsCount, maximumNumberOfWords);
|
||||
break;
|
||||
}
|
||||
|
||||
if (inGreeting == 1 && voiceDuration >= greeting) {
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "AMD: ANSWERING MACHINE: voiceDuration:%d greeting:%d\n", voiceDuration, greeting);
|
||||
ast_frfree(f);
|
||||
strcpy(amdStatus , "MACHINE");
|
||||
sprintf(amdCause , "LONGGREETING-%d-%d", voiceDuration, greeting);
|
||||
break;
|
||||
}
|
||||
|
||||
if (voiceDuration >= minimumWordLength ) {
|
||||
silenceDuration = 0;
|
||||
inInitialSilence = 0;
|
||||
inGreeting = 1;
|
||||
}
|
||||
|
||||
}
|
||||
}
|
||||
ast_frfree(f);
|
||||
}
|
||||
|
||||
if (!res) {
|
||||
/* It took too long to get a frame back. Giving up. */
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "AMD: Channel [%s]. Too long...\n", chan->name);
|
||||
strcpy(amdStatus , "NOTSURE");
|
||||
sprintf(amdCause , "TOOLONG-%d", iTotalTime);
|
||||
}
|
||||
|
||||
/* Set the status and cause on the channel */
|
||||
pbx_builtin_setvar_helper(chan , "AMDSTATUS" , amdStatus);
|
||||
pbx_builtin_setvar_helper(chan , "AMDCAUSE" , amdCause);
|
||||
|
||||
/* Restore channel read format */
|
||||
if (readFormat && ast_set_read_format(chan, readFormat))
|
||||
ast_log(LOG_WARNING, "AMD: Unable to restore read format on '%s'\n", chan->name);
|
||||
|
||||
/* Free the DSP used to detect silence */
|
||||
ast_dsp_free(silenceDetector);
|
||||
|
||||
return;
|
||||
}
|
||||
|
||||
|
||||
static int amd_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *u = NULL;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
isAnsweringMachine(chan, data);
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static void load_config(void)
|
||||
{
|
||||
struct ast_config *cfg = NULL;
|
||||
char *cat = NULL;
|
||||
struct ast_variable *var = NULL;
|
||||
|
||||
if (!(cfg = ast_config_load("amd.conf"))) {
|
||||
ast_log(LOG_ERROR, "Configuration file amd.conf missing.\n");
|
||||
return;
|
||||
}
|
||||
|
||||
cat = ast_category_browse(cfg, NULL);
|
||||
|
||||
while (cat) {
|
||||
if (!strcasecmp(cat, "general") ) {
|
||||
var = ast_variable_browse(cfg, cat);
|
||||
while (var) {
|
||||
if (!strcasecmp(var->name, "initial_silence")) {
|
||||
dfltInitialSilence = atoi(var->value);
|
||||
} else if (!strcasecmp(var->name, "greeting")) {
|
||||
dfltGreeting = atoi(var->value);
|
||||
} else if (!strcasecmp(var->name, "after_greeting_silence")) {
|
||||
dfltAfterGreetingSilence = atoi(var->value);
|
||||
} else if (!strcasecmp(var->name, "silence_threshold")) {
|
||||
dfltSilenceThreshold = atoi(var->value);
|
||||
} else if (!strcasecmp(var->name, "total_analysis_time")) {
|
||||
dfltTotalAnalysisTime = atoi(var->value);
|
||||
} else if (!strcasecmp(var->name, "min_word_length")) {
|
||||
dfltMinimumWordLength = atoi(var->value);
|
||||
} else if (!strcasecmp(var->name, "between_words_silence")) {
|
||||
dfltBetweenWordsSilence = atoi(var->value);
|
||||
} else if (!strcasecmp(var->name, "maximum_number_of_words")) {
|
||||
dfltMaximumNumberOfWords = atoi(var->value);
|
||||
} else {
|
||||
ast_log(LOG_WARNING, "%s: Cat:%s. Unknown keyword %s at line %d of amd.conf\n",
|
||||
app, cat, var->name, var->lineno);
|
||||
}
|
||||
var = var->next;
|
||||
}
|
||||
}
|
||||
cat = ast_category_browse(cfg, cat);
|
||||
}
|
||||
|
||||
ast_config_destroy(cfg);
|
||||
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "AMD defaults: initialSilence [%d] greeting [%d] afterGreetingSilence [%d] "
|
||||
"totalAnalysisTime [%d] minimumWordLength [%d] betweenWordsSilence [%d] maximumNumberOfWords [%d] silenceThreshold [%d] \n",
|
||||
dfltInitialSilence, dfltGreeting, dfltAfterGreetingSilence, dfltTotalAnalysisTime,
|
||||
dfltMinimumWordLength, dfltBetweenWordsSilence, dfltMaximumNumberOfWords, dfltSilenceThreshold );
|
||||
|
||||
return;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
ast_module_user_hangup_all();
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
load_config();
|
||||
return ast_register_application(app, amd_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
static int reload(void)
|
||||
{
|
||||
load_config();
|
||||
return 0;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Answering Machine Detection Application",
|
||||
.load = load_module,
|
||||
.unload = unload_module,
|
||||
.reload = reload,
|
||||
);
|
||||
@@ -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 Execute arbitrary authenticate commands
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
#include <errno.h>
|
||||
#include <stdio.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/astdb.h"
|
||||
#include "asterisk/utils.h"
|
||||
#include "asterisk/options.h"
|
||||
|
||||
enum {
|
||||
OPT_ACCOUNT = (1 << 0),
|
||||
OPT_DATABASE = (1 << 1),
|
||||
OPT_JUMP = (1 << 2),
|
||||
OPT_MULTIPLE = (1 << 3),
|
||||
OPT_REMOVE = (1 << 4),
|
||||
} auth_option_flags;
|
||||
|
||||
AST_APP_OPTIONS(auth_app_options, {
|
||||
AST_APP_OPTION('a', OPT_ACCOUNT),
|
||||
AST_APP_OPTION('d', OPT_DATABASE),
|
||||
AST_APP_OPTION('j', OPT_JUMP),
|
||||
AST_APP_OPTION('m', OPT_MULTIPLE),
|
||||
AST_APP_OPTION('r', OPT_REMOVE),
|
||||
});
|
||||
|
||||
|
||||
static char *app = "Authenticate";
|
||||
|
||||
static char *synopsis = "Authenticate a user";
|
||||
|
||||
static char *descrip =
|
||||
" Authenticate(password[|options[|maxdigits]]): This application asks the caller\n"
|
||||
"to enter a given password in order to continue dialplan execution. If the password\n"
|
||||
"begins with the '/' character, it is interpreted as a file which contains a list of\n"
|
||||
"valid passwords, listed 1 password per line in the file.\n"
|
||||
" When using a database key, the value associated with the key can be anything.\n"
|
||||
"Users have three attempts to authenticate before the channel is hung up. If the\n"
|
||||
"passsword is invalid, the 'j' option is specified, and priority n+101 exists,\n"
|
||||
"dialplan execution will continnue at this location.\n"
|
||||
" Options:\n"
|
||||
" a - Set the channels' account code to the password that is entered\n"
|
||||
" d - Interpret the given path as database key, not a literal file\n"
|
||||
" j - Support jumping to n+101 if authentication fails\n"
|
||||
" m - Interpret the given path as a file which contains a list of account\n"
|
||||
" codes and password hashes delimited with ':', listed one per line in\n"
|
||||
" the file. When one of the passwords is matched, the channel will have\n"
|
||||
" its account code set to the corresponding account code in the file.\n"
|
||||
" r - Remove the database key upon successful entry (valid with 'd' only)\n"
|
||||
" maxdigits - maximum acceptable number of digits. Stops reading after\n"
|
||||
" maxdigits have been entered (without requiring the user to\n"
|
||||
" press the '#' key).\n"
|
||||
" Defaults to 0 - no limit - wait for the user press the '#' key.\n"
|
||||
;
|
||||
|
||||
static int auth_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res=0;
|
||||
int retries;
|
||||
struct ast_module_user *u;
|
||||
char passwd[256];
|
||||
char *prompt;
|
||||
int maxdigits;
|
||||
char *argcopy =NULL;
|
||||
struct ast_flags flags = {0};
|
||||
|
||||
AST_DECLARE_APP_ARGS(arglist,
|
||||
AST_APP_ARG(password);
|
||||
AST_APP_ARG(options);
|
||||
AST_APP_ARG(maxdigits);
|
||||
);
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "Authenticate requires an argument(password)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (chan->_state != AST_STATE_UP) {
|
||||
res = ast_answer(chan);
|
||||
if (res) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
|
||||
argcopy = ast_strdupa(data);
|
||||
|
||||
AST_STANDARD_APP_ARGS(arglist,argcopy);
|
||||
|
||||
if (!ast_strlen_zero(arglist.options)) {
|
||||
ast_app_parse_options(auth_app_options, &flags, NULL, arglist.options);
|
||||
}
|
||||
|
||||
if (!ast_strlen_zero(arglist.maxdigits)) {
|
||||
maxdigits = atoi(arglist.maxdigits);
|
||||
if ((maxdigits<1) || (maxdigits>sizeof(passwd)-2))
|
||||
maxdigits = sizeof(passwd) - 2;
|
||||
} else {
|
||||
maxdigits = sizeof(passwd) - 2;
|
||||
}
|
||||
|
||||
/* Start asking for password */
|
||||
prompt = "agent-pass";
|
||||
for (retries = 0; retries < 3; retries++) {
|
||||
res = ast_app_getdata(chan, prompt, passwd, maxdigits, 0);
|
||||
if (res < 0)
|
||||
break;
|
||||
res = 0;
|
||||
if (arglist.password[0] == '/') {
|
||||
if (ast_test_flag(&flags,OPT_DATABASE)) {
|
||||
char tmp[256];
|
||||
/* Compare against a database key */
|
||||
if (!ast_db_get(arglist.password + 1, passwd, tmp, sizeof(tmp))) {
|
||||
/* It's a good password */
|
||||
if (ast_test_flag(&flags,OPT_REMOVE)) {
|
||||
ast_db_del(arglist.password + 1, passwd);
|
||||
}
|
||||
break;
|
||||
}
|
||||
} else {
|
||||
/* Compare against a file */
|
||||
FILE *f;
|
||||
f = fopen(arglist.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 (ast_test_flag(&flags,OPT_MULTIPLE)) {
|
||||
md5secret = strchr(buf, ':');
|
||||
if (md5secret == NULL)
|
||||
continue;
|
||||
*md5secret = '\0';
|
||||
md5secret++;
|
||||
ast_md5_hash(md5passwd, passwd);
|
||||
if (!strcmp(md5passwd, md5secret)) {
|
||||
if (ast_test_flag(&flags,OPT_ACCOUNT))
|
||||
ast_cdr_setaccount(chan, buf);
|
||||
break;
|
||||
}
|
||||
} else {
|
||||
if (!strcmp(passwd, buf)) {
|
||||
if (ast_test_flag(&flags,OPT_ACCOUNT))
|
||||
ast_cdr_setaccount(chan, buf);
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
fclose(f);
|
||||
if (!ast_strlen_zero(buf)) {
|
||||
if (ast_test_flag(&flags,OPT_MULTIPLE)) {
|
||||
if (md5secret && !strcmp(md5passwd, md5secret))
|
||||
break;
|
||||
} else {
|
||||
if (!strcmp(passwd, buf))
|
||||
break;
|
||||
}
|
||||
}
|
||||
} else
|
||||
ast_log(LOG_WARNING, "Unable to open file '%s' for authentication: %s\n", arglist.password, strerror(errno));
|
||||
}
|
||||
} else {
|
||||
/* Compare against a fixed password */
|
||||
if (!strcmp(passwd, arglist.password))
|
||||
break;
|
||||
}
|
||||
prompt="auth-incorrect";
|
||||
}
|
||||
if ((retries < 3) && !res) {
|
||||
if (ast_test_flag(&flags,OPT_ACCOUNT) && !ast_test_flag(&flags,OPT_MULTIPLE))
|
||||
ast_cdr_setaccount(chan, passwd);
|
||||
res = ast_streamfile(chan, "auth-thankyou", chan->language);
|
||||
if (!res)
|
||||
res = ast_waitstream(chan, "");
|
||||
} else {
|
||||
if (ast_test_flag(&flags,OPT_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;
|
||||
}
|
||||
}
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, auth_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Authentication Application");
|
||||
@@ -1,78 +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
|
||||
*
|
||||
* Martin Pycko <martinp@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <sys/types.h>
|
||||
#include <stdlib.h>
|
||||
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/pbx.h"
|
||||
|
||||
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";
|
||||
|
||||
|
||||
static int nocdr_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *u;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (chan->cdr) {
|
||||
ast_set_flag(chan->cdr, AST_CDR_FLAG_POST_DISABLED);
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(nocdr_app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(nocdr_app, nocdr_exec, nocdr_synopsis, nocdr_descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Tell Asterisk to not maintain a CDR for the current call");
|
||||
@@ -1,173 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
* \author James Golovich <james@gnuinter.net>
|
||||
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
#include <errno.h>
|
||||
#include <sys/ioctl.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/devicestate.h"
|
||||
#include "asterisk/options.h"
|
||||
|
||||
static char *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";
|
||||
|
||||
|
||||
static int chanavail_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res=-1, inuse=-1, option_state=0, priority_jump=0;
|
||||
int status;
|
||||
struct ast_module_user *u;
|
||||
char *info, tmp[512], trychan[512], *peers, *tech, *number, *rest, *cur;
|
||||
struct ast_channel *tempchan;
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(reqchans);
|
||||
AST_APP_ARG(options);
|
||||
);
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "ChanIsAvail requires an argument (Zap/1&Zap/2)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
info = ast_strdupa(data);
|
||||
|
||||
AST_STANDARD_APP_ARGS(args, info);
|
||||
|
||||
if (args.options) {
|
||||
if (strchr(args.options, 's'))
|
||||
option_state = 1;
|
||||
if (strchr(args.options, 'j'))
|
||||
priority_jump = 1;
|
||||
}
|
||||
peers = args.reqchans;
|
||||
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");
|
||||
ast_module_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 || ast_opt_priority_jumping) {
|
||||
if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res = 0;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, chanavail_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Check channel availability");
|
||||
@@ -1,140 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 2006, Sergey Basmanov
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief ChannelRedirect application
|
||||
*
|
||||
* \author Sergey Basmanov <sergey_basmanov@mail.ru>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/features.h"
|
||||
#include "asterisk/options.h"
|
||||
|
||||
static char *app = "ChannelRedirect";
|
||||
static char *synopsis = "Redirects given channel to a dialplan target.";
|
||||
static char *descrip =
|
||||
"ChannelRedirect(channel|[[context|]extension|]priority):\n"
|
||||
" Sends the specified channel to the specified extension priority\n";
|
||||
|
||||
|
||||
static int asyncgoto_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = -1;
|
||||
struct ast_module_user *u;
|
||||
char *info, *context, *exten, *priority;
|
||||
int prio = 1;
|
||||
struct ast_channel *chan2 = NULL;
|
||||
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(channel);
|
||||
AST_APP_ARG(label);
|
||||
);
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "%s requires an argument (channel|[[context|]exten|]priority)\n", app);
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
info = ast_strdupa(data);
|
||||
AST_STANDARD_APP_ARGS(args, info);
|
||||
|
||||
if (ast_strlen_zero(args.channel) || ast_strlen_zero(args.label)) {
|
||||
ast_log(LOG_WARNING, "%s requires an argument (channel|[[context|]exten|]priority)\n", app);
|
||||
goto quit;
|
||||
}
|
||||
|
||||
chan2 = ast_get_channel_by_name_locked(args.channel);
|
||||
if (!chan2) {
|
||||
ast_log(LOG_WARNING, "No such channel: %s\n", args.channel);
|
||||
goto quit;
|
||||
}
|
||||
|
||||
/* Parsed right to left, so standard parsing won't work */
|
||||
context = strsep(&args.label, "|");
|
||||
exten = strsep(&args.label, "|");
|
||||
if (exten) {
|
||||
priority = strsep(&args.label, "|");
|
||||
if (!priority) {
|
||||
priority = exten;
|
||||
exten = context;
|
||||
context = NULL;
|
||||
}
|
||||
} else {
|
||||
priority = context;
|
||||
context = NULL;
|
||||
}
|
||||
|
||||
/* ast_findlabel_extension does not convert numeric priorities; it only does a lookup */
|
||||
if (!(prio = atoi(priority)) && !(prio = ast_findlabel_extension(chan2, S_OR(context, chan2->context),
|
||||
S_OR(exten, chan2->exten), priority, chan2->cid.cid_num))) {
|
||||
ast_log(LOG_WARNING, "'%s' is not a known priority or label\n", priority);
|
||||
goto chanquit;
|
||||
}
|
||||
|
||||
if (option_debug > 1)
|
||||
ast_log(LOG_DEBUG, "Attempting async goto (%s) to %s|%s|%d\n", args.channel, S_OR(context, chan2->context), S_OR(exten, chan2->exten), prio);
|
||||
|
||||
if (ast_async_goto_if_exists(chan2, S_OR(context, chan2->context), S_OR(exten, chan2->exten), prio))
|
||||
ast_log(LOG_WARNING, "%s failed for %s\n", app, args.channel);
|
||||
else
|
||||
res = 0;
|
||||
|
||||
chanquit:
|
||||
ast_mutex_unlock(&chan2->lock);
|
||||
quit:
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, asyncgoto_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Channel Redirect");
|
||||
@@ -1,745 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 2005 Anthony Minessale II (anthmct@yahoo.com)
|
||||
* Copyright (C) 2005 - 2006, Digium, Inc.
|
||||
*
|
||||
* A license has been granted to Digium (via disclaimer) for the use of
|
||||
* this code.
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief ChanSpy: Listen in on any channel.
|
||||
*
|
||||
* \author Anthony Minessale II <anthmct@yahoo.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
#include <ctype.h>
|
||||
|
||||
#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"
|
||||
|
||||
#define AST_NAME_STRLEN 256
|
||||
|
||||
static const char *tdesc = "Listen to a channel, and optionally whisper into it";
|
||||
static const char *app_chan = "ChanSpy";
|
||||
static const char *desc_chan =
|
||||
" ChanSpy([chanprefix][|options]): This application is used to listen to the\n"
|
||||
"audio from an Asterisk channel. This includes the audio coming in and\n"
|
||||
"out of the channel being spied on. If the 'chanprefix' parameter is specified,\n"
|
||||
"only channels beginning with this string will be spied upon.\n"
|
||||
" While spying, the following actions may be performed:\n"
|
||||
" - Dialing # cycles the volume level.\n"
|
||||
" - Dialing * will stop spying and look for another channel to spy on.\n"
|
||||
" - Dialing a series of digits followed by # builds a channel name to append\n"
|
||||
" to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing\n"
|
||||
" the digits '1234#' while spying will begin spying on the channel\n"
|
||||
" 'Agent/1234'.\n"
|
||||
" Options:\n"
|
||||
" b - Only spy on channels involved in a bridged call.\n"
|
||||
" g(grp) - Match only channels where their ${SPYGROUP} variable is set to\n"
|
||||
" contain 'grp' in an optional : delimited list.\n"
|
||||
" q - Don't play a beep when beginning to spy on a channel, or speak the\n"
|
||||
" selected channel name.\n"
|
||||
" r[(basename)] - Record the session to the monitor spool directory. An\n"
|
||||
" optional base for the filename may be specified. The\n"
|
||||
" default is 'chanspy'.\n"
|
||||
" v([value]) - Adjust the initial volume in the range from -4 to 4. A\n"
|
||||
" negative value refers to a quieter setting.\n"
|
||||
" w - Enable 'whisper' mode, so the spying channel can talk to\n"
|
||||
" the spied-on channel.\n"
|
||||
" W - Enable 'private whisper' mode, so the spying channel can\n"
|
||||
" talk to the spied-on channel but cannot listen to that\n"
|
||||
" channel.\n"
|
||||
;
|
||||
|
||||
static const char *app_ext = "ExtenSpy";
|
||||
static const char *desc_ext =
|
||||
" ExtenSpy(exten[@context][|options]): This application is used to listen to the\n"
|
||||
"audio from an Asterisk channel. This includes the audio coming in and\n"
|
||||
"out of the channel being spied on. Only channels created by outgoing calls for the\n"
|
||||
"specified extension will be selected for spying. If the optional context is not\n"
|
||||
"supplied, the current channel's context will be used.\n"
|
||||
" While spying, the following actions may be performed:\n"
|
||||
" - Dialing # cycles the volume level.\n"
|
||||
" - Dialing * will stop spying and look for another channel to spy on.\n"
|
||||
" Options:\n"
|
||||
" b - Only spy on channels involved in a bridged call.\n"
|
||||
" g(grp) - Match only channels where their ${SPYGROUP} variable is set to\n"
|
||||
" contain 'grp' in an optional : delimited list.\n"
|
||||
" q - Don't play a beep when beginning to spy on a channel, or speak the\n"
|
||||
" selected channel name.\n"
|
||||
" r[(basename)] - Record the session to the monitor spool directory. An\n"
|
||||
" optional base for the filename may be specified. The\n"
|
||||
" default is 'chanspy'.\n"
|
||||
" v([value]) - Adjust the initial volume in the range from -4 to 4. A\n"
|
||||
" negative value refers to a quieter setting.\n"
|
||||
" w - Enable 'whisper' mode, so the spying channel can talk to\n"
|
||||
" the spied-on channel.\n"
|
||||
" W - Enable 'private whisper' mode, so the spying channel can\n"
|
||||
" talk to the spied-on channel but cannot listen to that\n"
|
||||
" channel.\n"
|
||||
;
|
||||
|
||||
enum {
|
||||
OPTION_QUIET = (1 << 0), /* Quiet, no announcement */
|
||||
OPTION_BRIDGED = (1 << 1), /* Only look at bridged calls */
|
||||
OPTION_VOLUME = (1 << 2), /* Specify initial volume */
|
||||
OPTION_GROUP = (1 << 3), /* Only look at channels in group */
|
||||
OPTION_RECORD = (1 << 4),
|
||||
OPTION_WHISPER = (1 << 5),
|
||||
OPTION_PRIVATE = (1 << 6), /* Private Whisper mode */
|
||||
} chanspy_opt_flags;
|
||||
|
||||
enum {
|
||||
OPT_ARG_VOLUME = 0,
|
||||
OPT_ARG_GROUP,
|
||||
OPT_ARG_RECORD,
|
||||
OPT_ARG_ARRAY_SIZE,
|
||||
} chanspy_opt_args;
|
||||
|
||||
AST_APP_OPTIONS(spy_opts, {
|
||||
AST_APP_OPTION('q', OPTION_QUIET),
|
||||
AST_APP_OPTION('b', OPTION_BRIDGED),
|
||||
AST_APP_OPTION('w', OPTION_WHISPER),
|
||||
AST_APP_OPTION('W', OPTION_PRIVATE),
|
||||
AST_APP_OPTION_ARG('v', OPTION_VOLUME, OPT_ARG_VOLUME),
|
||||
AST_APP_OPTION_ARG('g', OPTION_GROUP, OPT_ARG_GROUP),
|
||||
AST_APP_OPTION_ARG('r', OPTION_RECORD, OPT_ARG_RECORD),
|
||||
});
|
||||
|
||||
|
||||
struct chanspy_translation_helper {
|
||||
/* spy data */
|
||||
struct ast_channel_spy spy;
|
||||
int fd;
|
||||
int volfactor;
|
||||
};
|
||||
|
||||
static void *spy_alloc(struct ast_channel *chan, void *data)
|
||||
{
|
||||
/* just store the data pointer in the channel structure */
|
||||
return data;
|
||||
}
|
||||
|
||||
static void spy_release(struct ast_channel *chan, void *data)
|
||||
{
|
||||
/* nothing to do */
|
||||
}
|
||||
|
||||
static int spy_generate(struct ast_channel *chan, void *data, int len, int samples)
|
||||
{
|
||||
struct chanspy_translation_helper *csth = data;
|
||||
struct ast_frame *f;
|
||||
|
||||
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_channel_lock(chan);
|
||||
res = ast_channel_spy_add(chan, spy);
|
||||
ast_channel_unlock(chan);
|
||||
|
||||
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;
|
||||
csth->spy.read_vol_adjustment = csth->volfactor;
|
||||
csth->spy.write_vol_adjustment = csth->volfactor;
|
||||
}
|
||||
|
||||
static int channel_spy(struct ast_channel *chan, struct ast_channel *spyee, int *volfactor, int fd,
|
||||
const struct ast_flags *flags)
|
||||
{
|
||||
struct chanspy_translation_helper csth;
|
||||
int running = 0, res, x = 0;
|
||||
char inp[24] = {0};
|
||||
char *name;
|
||||
struct ast_frame *f;
|
||||
struct ast_silence_generator *silgen = NULL;
|
||||
|
||||
if (ast_check_hangup(chan) || ast_check_hangup(spyee))
|
||||
return 0;
|
||||
|
||||
name = ast_strdupa(spyee->name);
|
||||
if (option_verbose >= 2)
|
||||
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";
|
||||
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_mutex_destroy(&csth.spy.lock);
|
||||
return 0;
|
||||
}
|
||||
|
||||
if (ast_test_flag(flags, OPTION_WHISPER)) {
|
||||
struct ast_filestream *beepstream;
|
||||
int old_write_format = 0;
|
||||
|
||||
ast_channel_whisper_start(csth.spy.chan);
|
||||
old_write_format = chan->writeformat;
|
||||
if ((beepstream = ast_openstream_full(chan, "beep", chan->language, 1))) {
|
||||
struct ast_frame *f;
|
||||
|
||||
while ((f = ast_readframe(beepstream))) {
|
||||
ast_channel_whisper_feed(csth.spy.chan, f);
|
||||
ast_frfree(f);
|
||||
}
|
||||
|
||||
ast_closestream(beepstream);
|
||||
chan->stream = NULL;
|
||||
}
|
||||
if (old_write_format)
|
||||
ast_set_write_format(chan, old_write_format);
|
||||
}
|
||||
|
||||
if (ast_test_flag(flags, OPTION_PRIVATE))
|
||||
silgen = ast_channel_start_silence_generator(chan);
|
||||
else
|
||||
ast_activate_generator(chan, &spygen, &csth);
|
||||
|
||||
/* We can no longer rely on 'spyee' being an actual channel;
|
||||
it can be hung up and freed out from under us. However, the
|
||||
channel destructor will put NULL into our csth.spy.chan
|
||||
field when that happens, so that is our signal that the spyee
|
||||
channel has gone away.
|
||||
*/
|
||||
|
||||
/* Note: it is very important that the ast_waitfor() be the first
|
||||
condition in this expression, so that if we wait for some period
|
||||
of time before receiving a frame from our spying channel, we check
|
||||
for hangup on the spied-on channel _after_ knowing that a frame
|
||||
has arrived, since the spied-on channel could have gone away while
|
||||
we were waiting
|
||||
*/
|
||||
while ((res = ast_waitfor(chan, -1) > -1) &&
|
||||
csth.spy.status == CHANSPY_RUNNING &&
|
||||
csth.spy.chan) {
|
||||
if (!(f = ast_read(chan)) || ast_check_hangup(chan)) {
|
||||
running = -1;
|
||||
break;
|
||||
}
|
||||
|
||||
if (ast_test_flag(flags, OPTION_WHISPER) &&
|
||||
(f->frametype == AST_FRAME_VOICE)) {
|
||||
ast_channel_whisper_feed(csth.spy.chan, f);
|
||||
ast_frfree(f);
|
||||
continue;
|
||||
}
|
||||
|
||||
res = (f->frametype == AST_FRAME_DTMF) ? f->subclass : 0;
|
||||
ast_frfree(f);
|
||||
if (!res)
|
||||
continue;
|
||||
|
||||
if (x == sizeof(inp))
|
||||
x = 0;
|
||||
|
||||
if (res < 0) {
|
||||
running = -1;
|
||||
break;
|
||||
}
|
||||
|
||||
if (res == '*') {
|
||||
running = 0;
|
||||
break;
|
||||
} else if (res == '#') {
|
||||
if (!ast_strlen_zero(inp)) {
|
||||
running = atoi(inp);
|
||||
break;
|
||||
}
|
||||
|
||||
(*volfactor)++;
|
||||
if (*volfactor > 4)
|
||||
*volfactor = -4;
|
||||
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 >= '0' && res <= '9') {
|
||||
inp[x++] = res;
|
||||
}
|
||||
}
|
||||
|
||||
if (ast_test_flag(flags, OPTION_WHISPER) && csth.spy.chan)
|
||||
ast_channel_whisper_stop(csth.spy.chan);
|
||||
|
||||
if (ast_test_flag(flags, OPTION_PRIVATE))
|
||||
ast_channel_stop_silence_generator(chan, silgen);
|
||||
else
|
||||
ast_deactivate_generator(chan);
|
||||
|
||||
/* If a channel still exists on our spy structure then we need to remove ourselves */
|
||||
if (csth.spy.chan) {
|
||||
csth.spy.status = CHANSPY_DONE;
|
||||
ast_channel_lock(csth.spy.chan);
|
||||
ast_channel_spy_remove(csth.spy.chan, &csth.spy);
|
||||
ast_channel_unlock(csth.spy.chan);
|
||||
}
|
||||
ast_channel_spy_free(&csth.spy);
|
||||
|
||||
if (option_verbose >= 2)
|
||||
ast_verbose(VERBOSE_PREFIX_2 "Done Spying on channel %s\n", name);
|
||||
|
||||
return running;
|
||||
}
|
||||
|
||||
static struct ast_channel *next_channel(const struct ast_channel *last, const char *spec,
|
||||
const char *exten, const char *context)
|
||||
{
|
||||
struct ast_channel *this;
|
||||
|
||||
redo:
|
||||
if (spec)
|
||||
this = ast_walk_channel_by_name_prefix_locked(last, spec, strlen(spec));
|
||||
else if (exten)
|
||||
this = ast_walk_channel_by_exten_locked(last, exten, context);
|
||||
else
|
||||
this = ast_channel_walk_locked(last);
|
||||
|
||||
if (this) {
|
||||
ast_channel_unlock(this);
|
||||
if (!strncmp(this->name, "Zap/pseudo", 10))
|
||||
goto redo;
|
||||
}
|
||||
|
||||
return this;
|
||||
}
|
||||
|
||||
static int common_exec(struct ast_channel *chan, const struct ast_flags *flags,
|
||||
int volfactor, const int fd, const char *mygroup, const char *spec,
|
||||
const char *exten, const char *context)
|
||||
{
|
||||
struct ast_channel *peer, *prev, *next;
|
||||
char nameprefix[AST_NAME_STRLEN];
|
||||
char peer_name[AST_NAME_STRLEN + 5];
|
||||
signed char zero_volume = 0;
|
||||
int waitms;
|
||||
int res;
|
||||
char *ptr;
|
||||
int num;
|
||||
|
||||
if (chan->_state != AST_STATE_UP)
|
||||
ast_answer(chan);
|
||||
|
||||
ast_set_flag(chan, AST_FLAG_SPYING); /* so nobody can spy on us while we are spying */
|
||||
|
||||
waitms = 100;
|
||||
|
||||
for (;;) {
|
||||
if (!ast_test_flag(flags, OPTION_QUIET)) {
|
||||
res = ast_streamfile(chan, "beep", chan->language);
|
||||
if (!res)
|
||||
res = ast_waitstream(chan, "");
|
||||
else if (res < 0) {
|
||||
ast_clear_flag(chan, AST_FLAG_SPYING);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
res = ast_waitfordigit(chan, waitms);
|
||||
if (res < 0) {
|
||||
ast_clear_flag(chan, AST_FLAG_SPYING);
|
||||
break;
|
||||
}
|
||||
|
||||
/* reset for the next loop around, unless overridden later */
|
||||
waitms = 100;
|
||||
peer = prev = next = NULL;
|
||||
|
||||
for (peer = next_channel(peer, spec, exten, context);
|
||||
peer;
|
||||
prev = peer, peer = next ? next : next_channel(peer, spec, exten, context), next = NULL) {
|
||||
const char *group;
|
||||
int igrp = !mygroup;
|
||||
char *groups[25];
|
||||
int num_groups = 0;
|
||||
char *dup_group;
|
||||
int x;
|
||||
char *s;
|
||||
|
||||
if (peer == prev)
|
||||
break;
|
||||
|
||||
if (peer == chan)
|
||||
continue;
|
||||
|
||||
if (ast_test_flag(flags, OPTION_BRIDGED) && !ast_bridged_channel(peer))
|
||||
continue;
|
||||
|
||||
if (ast_check_hangup(peer) || ast_test_flag(peer, AST_FLAG_SPYING))
|
||||
continue;
|
||||
|
||||
if (mygroup) {
|
||||
if ((group = pbx_builtin_getvar_helper(peer, "SPYGROUP"))) {
|
||||
dup_group = ast_strdupa(group);
|
||||
num_groups = ast_app_separate_args(dup_group, ':', groups,
|
||||
sizeof(groups) / sizeof(groups[0]));
|
||||
}
|
||||
|
||||
for (x = 0; x < num_groups; x++) {
|
||||
if (!strcmp(mygroup, groups[x])) {
|
||||
igrp = 1;
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
if (!igrp)
|
||||
continue;
|
||||
|
||||
strcpy(peer_name, "spy-");
|
||||
strncat(peer_name, peer->name, AST_NAME_STRLEN);
|
||||
ptr = strchr(peer_name, '/');
|
||||
*ptr++ = '\0';
|
||||
|
||||
for (s = peer_name; s < ptr; s++)
|
||||
*s = tolower(*s);
|
||||
|
||||
if (!ast_test_flag(flags, OPTION_QUIET)) {
|
||||
if (ast_fileexists(peer_name, NULL, NULL) != -1) {
|
||||
res = ast_streamfile(chan, peer_name, chan->language);
|
||||
if (!res)
|
||||
res = ast_waitstream(chan, "");
|
||||
if (res)
|
||||
break;
|
||||
} else
|
||||
res = ast_say_character_str(chan, peer_name, "", chan->language);
|
||||
if ((num = atoi(ptr)))
|
||||
ast_say_digits(chan, atoi(ptr), "", chan->language);
|
||||
}
|
||||
|
||||
waitms = 5000;
|
||||
res = channel_spy(chan, peer, &volfactor, fd, flags);
|
||||
|
||||
if (res == -1) {
|
||||
break;
|
||||
} else if (res > 1 && spec) {
|
||||
snprintf(nameprefix, AST_NAME_STRLEN, "%s/%d", spec, res);
|
||||
if ((next = ast_get_channel_by_name_prefix_locked(nameprefix, strlen(nameprefix)))) {
|
||||
ast_channel_unlock(next);
|
||||
} else {
|
||||
/* stay on this channel */
|
||||
next = peer;
|
||||
}
|
||||
peer = NULL;
|
||||
}
|
||||
}
|
||||
if (res == -1)
|
||||
break;
|
||||
}
|
||||
|
||||
ast_clear_flag(chan, AST_FLAG_SPYING);
|
||||
|
||||
ast_channel_setoption(chan, AST_OPTION_TXGAIN, &zero_volume, sizeof(zero_volume), 0);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int chanspy_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *u;
|
||||
char *options = NULL;
|
||||
char *spec = NULL;
|
||||
char *argv[2];
|
||||
char *mygroup = NULL;
|
||||
char *recbase = NULL;
|
||||
int fd = 0;
|
||||
struct ast_flags flags;
|
||||
int oldwf = 0;
|
||||
int argc = 0;
|
||||
int volfactor = 0;
|
||||
int res;
|
||||
|
||||
data = ast_strdupa(data);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if ((argc = ast_app_separate_args(data, '|', 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(spy_opts, &flags, opts, options);
|
||||
if (ast_test_flag(&flags, OPTION_GROUP))
|
||||
mygroup = opts[OPT_ARG_GROUP];
|
||||
|
||||
if (ast_test_flag(&flags, OPTION_RECORD) &&
|
||||
!(recbase = opts[OPT_ARG_RECORD]))
|
||||
recbase = "chanspy";
|
||||
|
||||
if (ast_test_flag(&flags, OPTION_VOLUME) && opts[OPT_ARG_VOLUME]) {
|
||||
int vol;
|
||||
|
||||
if ((sscanf(opts[OPT_ARG_VOLUME], "%d", &vol) != 1) || (vol > 4) || (vol < -4))
|
||||
ast_log(LOG_NOTICE, "Volume factor must be a number between -4 and 4\n");
|
||||
else
|
||||
volfactor = vol;
|
||||
}
|
||||
|
||||
if (ast_test_flag(&flags, OPTION_PRIVATE))
|
||||
ast_set_flag(&flags, OPTION_WHISPER);
|
||||
}
|
||||
|
||||
oldwf = chan->writeformat;
|
||||
if (ast_set_write_format(chan, AST_FORMAT_SLINEAR) < 0) {
|
||||
ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
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;
|
||||
}
|
||||
}
|
||||
|
||||
res = common_exec(chan, &flags, volfactor, fd, mygroup, spec, NULL, NULL);
|
||||
|
||||
if (fd)
|
||||
close(fd);
|
||||
|
||||
if (oldwf && ast_set_write_format(chan, oldwf) < 0)
|
||||
ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int extenspy_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *u;
|
||||
char *options = NULL;
|
||||
char *exten = NULL;
|
||||
char *context = NULL;
|
||||
char *argv[2];
|
||||
char *mygroup = NULL;
|
||||
char *recbase = NULL;
|
||||
int fd = 0;
|
||||
struct ast_flags flags;
|
||||
int oldwf = 0;
|
||||
int argc = 0;
|
||||
int volfactor = 0;
|
||||
int res;
|
||||
|
||||
data = ast_strdupa(data);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if ((argc = ast_app_separate_args(data, '|', argv, sizeof(argv) / sizeof(argv[0])))) {
|
||||
context = argv[0];
|
||||
if (!ast_strlen_zero(argv[0]))
|
||||
exten = strsep(&context, "@");
|
||||
if (ast_strlen_zero(context))
|
||||
context = ast_strdupa(chan->context);
|
||||
if (argc > 1)
|
||||
options = argv[1];
|
||||
}
|
||||
|
||||
if (options) {
|
||||
char *opts[OPT_ARG_ARRAY_SIZE];
|
||||
|
||||
ast_app_parse_options(spy_opts, &flags, opts, options);
|
||||
if (ast_test_flag(&flags, OPTION_GROUP))
|
||||
mygroup = opts[OPT_ARG_GROUP];
|
||||
|
||||
if (ast_test_flag(&flags, OPTION_RECORD) &&
|
||||
!(recbase = opts[OPT_ARG_RECORD]))
|
||||
recbase = "chanspy";
|
||||
|
||||
if (ast_test_flag(&flags, OPTION_VOLUME) && opts[OPT_ARG_VOLUME]) {
|
||||
int vol;
|
||||
|
||||
if ((sscanf(opts[OPT_ARG_VOLUME], "%d", &vol) != 1) || (vol > 4) || (vol < -4))
|
||||
ast_log(LOG_NOTICE, "Volume factor must be a number between -4 and 4\n");
|
||||
else
|
||||
volfactor = vol;
|
||||
}
|
||||
|
||||
if (ast_test_flag(&flags, OPTION_PRIVATE))
|
||||
ast_set_flag(&flags, OPTION_WHISPER);
|
||||
}
|
||||
|
||||
oldwf = chan->writeformat;
|
||||
if (ast_set_write_format(chan, AST_FORMAT_SLINEAR) < 0) {
|
||||
ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
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;
|
||||
}
|
||||
}
|
||||
|
||||
res = common_exec(chan, &flags, volfactor, fd, mygroup, NULL, exten, context);
|
||||
|
||||
if (fd)
|
||||
close(fd);
|
||||
|
||||
if (oldwf && ast_set_write_format(chan, oldwf) < 0)
|
||||
ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res = 0;
|
||||
|
||||
res |= ast_unregister_application(app_chan);
|
||||
res |= ast_unregister_application(app_ext);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
int res = 0;
|
||||
|
||||
res |= ast_register_application(app_chan, chanspy_exec, tdesc, desc_chan);
|
||||
res |= ast_register_application(app_ext, extenspy_exec, tdesc, desc_ext);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Listen to the audio of an active channel");
|
||||
@@ -1,168 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/translate.h"
|
||||
#include "asterisk/utils.h"
|
||||
#include "asterisk/options.h"
|
||||
|
||||
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";
|
||||
|
||||
|
||||
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 ast_module_user *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;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
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");
|
||||
ast_module_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 || ast_opt_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");
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
res = ast_unregister_application(app);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, controlplayback_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Control Playback Application");
|
||||
167
apps/app_db.c
167
apps/app_db.c
@@ -1,167 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
* \author Jefferson Noxon <jeff@debian.org>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
#include <sys/types.h>
|
||||
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/astdb.h"
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/options.h"
|
||||
|
||||
/*! \todo XXX Remove this application after 1.4 is relased */
|
||||
static char *d_descrip =
|
||||
" DBdel(family/key): This application will delete a key from the Asterisk\n"
|
||||
"database.\n"
|
||||
" This application has been DEPRECATED in favor of the DB_DELETE function.\n";
|
||||
|
||||
static char *dt_descrip =
|
||||
" DBdeltree(family[/keytree]): This application will delete a family or keytree\n"
|
||||
"from the Asterisk database\n";
|
||||
|
||||
static char *d_app = "DBdel";
|
||||
static char *dt_app = "DBdeltree";
|
||||
|
||||
static char *d_synopsis = "Delete a key from the database";
|
||||
static char *dt_synopsis = "Delete a family or keytree from the database";
|
||||
|
||||
|
||||
static int deltree_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
char *argv, *family, *keytree;
|
||||
struct ast_module_user *u;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
argv = ast_strdupa(data);
|
||||
|
||||
if (strchr(argv, '/')) {
|
||||
family = strsep(&argv, "/");
|
||||
keytree = strsep(&argv, "\0");
|
||||
if (!family || !keytree) {
|
||||
ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
|
||||
ast_module_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");
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int del_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
char *argv, *family, *key;
|
||||
struct ast_module_user *u;
|
||||
static int deprecation_warning = 0;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (!deprecation_warning) {
|
||||
deprecation_warning = 1;
|
||||
ast_log(LOG_WARNING, "The DBdel application has been deprecated in favor of the DB_DELETE dialplan function!\n");
|
||||
}
|
||||
|
||||
argv = ast_strdupa(data);
|
||||
|
||||
if (strchr(argv, '/')) {
|
||||
family = strsep(&argv, "/");
|
||||
key = strsep(&argv, "\0");
|
||||
if (!family || !key) {
|
||||
ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
|
||||
ast_module_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");
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int retval;
|
||||
|
||||
retval = ast_unregister_application(dt_app);
|
||||
retval |= ast_unregister_application(d_app);
|
||||
|
||||
return retval;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
int retval;
|
||||
|
||||
retval = ast_register_application(d_app, del_exec, d_synopsis, d_descrip);
|
||||
retval |= ast_register_application(dt_app, deltree_exec, dt_synopsis, dt_descrip);
|
||||
|
||||
return retval;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Database Access Functions");
|
||||
1970
apps/app_dial.c
1970
apps/app_dial.c
File diff suppressed because it is too large
Load Diff
@@ -1,349 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 2005, Anthony Minessale II
|
||||
*
|
||||
* Anthony Minessale II <anthmct@yahoo.com>
|
||||
*
|
||||
* Donated by Sangoma Technologies <http://www.samgoma.com>
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief Virtual Dictation Machine Application For Asterisk
|
||||
*
|
||||
* \author Anthony Minessale II <anthmct@yahoo.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
#include <sys/stat.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/say.h"
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/app.h"
|
||||
|
||||
static char *app = "Dictate";
|
||||
static char *synopsis = "Virtual Dictation Machine";
|
||||
static char *desc = " Dictate([<base_dir>[|<filename>]])\n"
|
||||
"Start dictation machine using optional base dir for files.\n";
|
||||
|
||||
|
||||
typedef enum {
|
||||
DFLAG_RECORD = (1 << 0),
|
||||
DFLAG_PLAY = (1 << 1),
|
||||
DFLAG_TRUNC = (1 << 2),
|
||||
DFLAG_PAUSE = (1 << 3),
|
||||
} dflags;
|
||||
|
||||
typedef enum {
|
||||
DMODE_INIT,
|
||||
DMODE_RECORD,
|
||||
DMODE_PLAY
|
||||
} dmodes;
|
||||
|
||||
#define ast_toggle_flag(it,flag) if(ast_test_flag(it, flag)) ast_clear_flag(it, flag); else ast_set_flag(it, flag)
|
||||
|
||||
static int play_and_wait(struct ast_channel *chan, char *file, char *digits)
|
||||
{
|
||||
int res = -1;
|
||||
if (!ast_streamfile(chan, file, chan->language)) {
|
||||
res = ast_waitstream(chan, digits);
|
||||
}
|
||||
return res;
|
||||
}
|
||||
|
||||
static int dictate_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
char *path = NULL, filein[256], *filename = "";
|
||||
char *parse;
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(base);
|
||||
AST_APP_ARG(filename);
|
||||
);
|
||||
char dftbase[256];
|
||||
char *base;
|
||||
struct ast_flags flags = {0};
|
||||
struct ast_filestream *fs;
|
||||
struct ast_frame *f = NULL;
|
||||
struct ast_module_user *u;
|
||||
int ffactor = 320 * 80,
|
||||
res = 0,
|
||||
done = 0,
|
||||
oldr = 0,
|
||||
lastop = 0,
|
||||
samples = 0,
|
||||
speed = 1,
|
||||
digit = 0,
|
||||
len = 0,
|
||||
maxlen = 0,
|
||||
mode = 0;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
snprintf(dftbase, sizeof(dftbase), "%s/dictate", ast_config_AST_SPOOL_DIR);
|
||||
if (!ast_strlen_zero(data)) {
|
||||
parse = ast_strdupa(data);
|
||||
AST_STANDARD_APP_ARGS(args, parse);
|
||||
} else
|
||||
args.argc = 0;
|
||||
|
||||
if (args.argc && !ast_strlen_zero(args.base)) {
|
||||
base = args.base;
|
||||
} else {
|
||||
base = dftbase;
|
||||
}
|
||||
if (args.argc > 1 && args.filename) {
|
||||
filename = args.filename;
|
||||
}
|
||||
oldr = chan->readformat;
|
||||
if ((res = ast_set_read_format(chan, AST_FORMAT_SLINEAR)) < 0) {
|
||||
ast_log(LOG_WARNING, "Unable to set to linear mode.\n");
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
ast_answer(chan);
|
||||
ast_safe_sleep(chan, 200);
|
||||
for (res = 0; !res;) {
|
||||
if (ast_strlen_zero(filename)) {
|
||||
if (ast_app_getdata(chan, "dictate/enter_filename", filein, sizeof(filein), 0) ||
|
||||
ast_strlen_zero(filein)) {
|
||||
res = -1;
|
||||
break;
|
||||
}
|
||||
} else {
|
||||
ast_copy_string(filein, filename, sizeof(filein));
|
||||
filename = "";
|
||||
}
|
||||
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);
|
||||
}
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
res = ast_unregister_application(app);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, dictate_exec, synopsis, desc);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Virtual Dictation Machine");
|
||||
@@ -1,181 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 2005, Joshua Colp
|
||||
*
|
||||
* Joshua Colp <jcolp@digium.com>
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief Directed Call Pickup Support
|
||||
*
|
||||
* \author Joshua Colp <jcolp@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/options.h"
|
||||
|
||||
#define PICKUPMARK "PICKUPMARK"
|
||||
|
||||
static const char *app = "Pickup";
|
||||
static const char *synopsis = "Directed Call Pickup";
|
||||
static const char *descrip =
|
||||
" Pickup(extension[@context][&extension2@context...]): This application can pickup any ringing channel\n"
|
||||
"that is calling the specified extension. If no context is specified, the current\n"
|
||||
"context will be used. If you use the special string \"PICKUPMARK\" for the context parameter, for example\n"
|
||||
"10@PICKUPMARK, this application tries to find a channel which has defined a channel variable with the same content\n"
|
||||
"as \"extension\".";
|
||||
|
||||
/* Perform actual pickup between two channels */
|
||||
static int pickup_do(struct ast_channel *chan, struct ast_channel *target)
|
||||
{
|
||||
int res = 0;
|
||||
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "Call pickup on '%s' by '%s'\n", target->name, chan->name);
|
||||
|
||||
if ((res = ast_answer(chan))) {
|
||||
ast_log(LOG_WARNING, "Unable to answer '%s'\n", chan->name);
|
||||
return -1;
|
||||
}
|
||||
|
||||
if ((res = ast_queue_control(chan, AST_CONTROL_ANSWER))) {
|
||||
ast_log(LOG_WARNING, "Unable to queue answer on '%s'\n", chan->name);
|
||||
return -1;
|
||||
}
|
||||
|
||||
if ((res = ast_channel_masquerade(target, chan))) {
|
||||
ast_log(LOG_WARNING, "Unable to masquerade '%s' into '%s'\n", chan->name, target->name);
|
||||
return -1;
|
||||
}
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
/* Helper function that determines whether a channel is capable of being picked up */
|
||||
static int can_pickup(struct ast_channel *chan)
|
||||
{
|
||||
if (!chan->pbx && (chan->_state == AST_STATE_RINGING || chan->_state == AST_STATE_RING))
|
||||
return 1;
|
||||
else
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* Attempt to pick up specified extension with context */
|
||||
static int pickup_by_exten(struct ast_channel *chan, char *exten, char *context)
|
||||
{
|
||||
int res = -1;
|
||||
struct ast_channel *target = NULL;
|
||||
|
||||
while ((target = ast_channel_walk_locked(target))) {
|
||||
if ((!strcasecmp(target->macroexten, exten) || !strcasecmp(target->exten, exten)) &&
|
||||
!strcasecmp(target->dialcontext, context) &&
|
||||
can_pickup(target)) {
|
||||
res = pickup_do(chan, target);
|
||||
ast_channel_unlock(target);
|
||||
break;
|
||||
}
|
||||
ast_channel_unlock(target);
|
||||
}
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
/* Attempt to pick up specified mark */
|
||||
static int pickup_by_mark(struct ast_channel *chan, char *mark)
|
||||
{
|
||||
int res = -1;
|
||||
const char *tmp = NULL;
|
||||
struct ast_channel *target = NULL;
|
||||
|
||||
while ((target = ast_channel_walk_locked(target))) {
|
||||
if ((tmp = pbx_builtin_getvar_helper(target, PICKUPMARK)) &&
|
||||
!strcasecmp(tmp, mark) &&
|
||||
can_pickup(target)) {
|
||||
res = pickup_do(chan, target);
|
||||
ast_channel_unlock(target);
|
||||
break;
|
||||
}
|
||||
ast_channel_unlock(target);
|
||||
}
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
/* Main application entry point */
|
||||
static int pickup_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *u = NULL;
|
||||
char *tmp = ast_strdupa(data);
|
||||
char *exten = NULL, *context = NULL;
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "Pickup requires an argument (extension)!\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
/* Parse extension (and context if there) */
|
||||
while (!ast_strlen_zero(tmp) && (exten = strsep(&tmp, "&"))) {
|
||||
if ((context = strchr(exten, '@')))
|
||||
*context++ = '\0';
|
||||
if (context && !strcasecmp(context, PICKUPMARK)) {
|
||||
if (!pickup_by_mark(chan, exten))
|
||||
break;
|
||||
} else {
|
||||
if (!pickup_by_exten(chan, exten, context ? context : chan->context))
|
||||
break;
|
||||
}
|
||||
ast_log(LOG_NOTICE, "No target channel found for %s.\n", exten);
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, pickup_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Directed Call Pickup Application");
|
||||
@@ -1,194 +1,59 @@
|
||||
/*
|
||||
* 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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
#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/config.h>
|
||||
#include <asterisk/say.h>
|
||||
#include <string.h>
|
||||
#include <ctype.h>
|
||||
#include <stdlib.h>
|
||||
#include <pthread.h>
|
||||
#include <stdio.h>
|
||||
#include "../asterisk.h"
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/config.h"
|
||||
#include "asterisk/say.h"
|
||||
#include "asterisk/utils.h"
|
||||
#include "asterisk/app.h"
|
||||
|
||||
#ifdef 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
|
||||
|
||||
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"
|
||||
" e - In addition to the name, also read the extension number to the\n"
|
||||
" caller before presenting dialing 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";
|
||||
" Directory(context): Presents the user with a directory of extensions from\n"
|
||||
"which they may select by name. The list of names and extensions is\n"
|
||||
"discovered from voicemail.conf. The context argument is required, and\n"
|
||||
"specifies the context in which to interpret the extensions\n. Returns 0\n"
|
||||
"unless the user hangs up. It also sets up the channel on exit to enter the\n"
|
||||
"extension the user selected.\n";
|
||||
|
||||
/* 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
|
||||
|
||||
STANDARD_LOCAL_USER;
|
||||
|
||||
#ifdef 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];
|
||||
struct odbc_obj *obj;
|
||||
LOCAL_USER_DECL;
|
||||
|
||||
obj = ast_odbc_request_obj(odbc_database, 1);
|
||||
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 = ast_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);
|
||||
ast_odbc_release_obj(obj);
|
||||
} 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(const char *lastname)
|
||||
static char *convert(char *lastname)
|
||||
{
|
||||
char *tmp;
|
||||
int lcount = 0;
|
||||
tmp = ast_malloc(NUMDIGITS + 1);
|
||||
tmp = malloc(NUMDIGITS + 1);
|
||||
if (tmp) {
|
||||
while((*lastname > 32) && lcount < NUMDIGITS) {
|
||||
switch(toupper(*lastname)) {
|
||||
@@ -245,6 +110,7 @@ static char *convert(const char *lastname)
|
||||
case 'Z':
|
||||
tmp[lcount++] = '9';
|
||||
break;
|
||||
default:
|
||||
}
|
||||
lastname++;
|
||||
}
|
||||
@@ -253,190 +119,19 @@ static char *convert(const 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 readext,
|
||||
int fromappvm)
|
||||
{
|
||||
int res = 0;
|
||||
int loop;
|
||||
char fn[256];
|
||||
|
||||
/* Check for the VoiceMail2 greeting first */
|
||||
snprintf(fn, sizeof(fn), "%s/voicemail/%s/%s/greet",
|
||||
ast_config_AST_SPOOL_DIR, context, ext);
|
||||
#ifdef ODBC_STORAGE
|
||||
retrieve_file(fn);
|
||||
#endif
|
||||
|
||||
if (ast_fileexists(fn, NULL, chan->language) <= 0) {
|
||||
/* no file, check for an old-style Voicemail greeting */
|
||||
snprintf(fn, sizeof(fn), "%s/vm/%s/greet",
|
||||
ast_config_AST_SPOOL_DIR, ext);
|
||||
}
|
||||
#ifdef ODBC_STORAGE
|
||||
retrieve_file(fn);
|
||||
#endif
|
||||
|
||||
if (ast_fileexists(fn, NULL, chan->language) > 0) {
|
||||
res = ast_stream_and_wait(chan, fn, chan->language, AST_DIGIT_ANY);
|
||||
ast_stopstream(chan);
|
||||
/* If Option 'e' was specified, also read the extension number with the name */
|
||||
if (readext) {
|
||||
ast_stream_and_wait(chan, "vm-extension", chan->language, AST_DIGIT_ANY);
|
||||
res = ast_say_character_str(chan, ext, AST_DIGIT_ANY, chan->language);
|
||||
}
|
||||
} else {
|
||||
res = ast_say_character_str(chan, S_OR(name, ext), AST_DIGIT_ANY, chan->language);
|
||||
if (!ast_strlen_zero(name) && readext) {
|
||||
ast_stream_and_wait(chan, "vm-extension", chan->language, AST_DIGIT_ANY);
|
||||
res = ast_say_character_str(chan, ext, AST_DIGIT_ANY, chan->language);
|
||||
}
|
||||
}
|
||||
#ifdef ODBC_STORAGE
|
||||
ast_filedelete(fn, NULL);
|
||||
#endif
|
||||
|
||||
for (loop = 3 ; loop > 0; loop--) {
|
||||
if (!res)
|
||||
res = ast_stream_and_wait(chan, "dir-instr", chan->language, AST_DIGIT_ANY);
|
||||
if (!res)
|
||||
res = ast_waitfordigit(chan, 3000);
|
||||
ast_stopstream(chan);
|
||||
|
||||
if (res < 0) /* User hungup, so jump out now */
|
||||
break;
|
||||
if (res == '1') { /* Name selected */
|
||||
if (fromappvm) {
|
||||
/* We still want to set the exten though */
|
||||
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;
|
||||
}
|
||||
if (res == '*') /* Skip to next match in list */
|
||||
break;
|
||||
|
||||
/* Not '1', or '*', so decrement number of tries */
|
||||
res = 0;
|
||||
}
|
||||
|
||||
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;
|
||||
const char *fullname;
|
||||
const 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 = NULL;
|
||||
while ( (mailbox = ast_category_browse(rtdata, 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);
|
||||
}
|
||||
ast_config_destroy(rtdata);
|
||||
|
||||
return cfg;
|
||||
}
|
||||
|
||||
static int do_directory(struct ast_channel *chan, struct ast_config *cfg, struct ast_config *ucfg, char *context, char *dialcontext, char digit, int last, int readext, 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], *cat;
|
||||
char name[80] = "";
|
||||
char ext[NUMDIGITS + 1];
|
||||
struct ast_variable *v;
|
||||
int res;
|
||||
int found=0;
|
||||
int lastuserchoice = 0;
|
||||
char *start, *conv, *stringp = NULL;
|
||||
const char *pos;
|
||||
|
||||
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;
|
||||
if (ast_readstring(chan, ext + 1, NUMDIGITS, 3000, 3000, "#") < 0) res = -1;
|
||||
if (!res) {
|
||||
/* Search for all names which start with those digits */
|
||||
v = ast_variable_browse(cfg, context);
|
||||
@@ -445,18 +140,16 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, struct
|
||||
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);
|
||||
@@ -470,95 +163,51 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, struct
|
||||
}
|
||||
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, readext, 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, AST_DIGIT_ANY, chan->language);
|
||||
}
|
||||
ahem:
|
||||
if (!res)
|
||||
res = ast_streamfile(chan, "dir-instr", chan->language);
|
||||
if (!res)
|
||||
res = ast_waitstream(chan, AST_DIGIT_ANY);
|
||||
if (!res)
|
||||
res = ast_waitfordigit(chan, 3000);
|
||||
ast_stopstream(chan);
|
||||
if (res > -1) {
|
||||
if (res == '1') {
|
||||
strncpy(chan->exten, v->name, sizeof(chan->exten)-1);
|
||||
chan->priority = 0;
|
||||
strncpy(chan->context, context, sizeof(chan->context)-1);
|
||||
res = 0;
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
v = v->next;
|
||||
}
|
||||
}
|
||||
|
||||
if (!res && ucfg) {
|
||||
/* Search users.conf for all names which start with those digits */
|
||||
for (cat = ast_category_browse(ucfg, NULL); cat && !res ; cat = ast_category_browse(ucfg, cat)) {
|
||||
if (!strcasecmp(cat, "general"))
|
||||
continue;
|
||||
if (!ast_true(ast_config_option(ucfg, cat, "hasdirectory")))
|
||||
continue;
|
||||
|
||||
/* Find all candidate extensions */
|
||||
if ((pos = ast_variable_retrieve(ucfg, cat, "fullname"))) {
|
||||
ast_copy_string(name, pos, sizeof(name));
|
||||
/* Grab the last name */
|
||||
if (last && strrchr(pos,' '))
|
||||
pos = strrchr(pos, ' ') + 1;
|
||||
conv = convert(pos);
|
||||
if (conv) {
|
||||
if (!strcmp(conv, ext)) {
|
||||
/* Match! */
|
||||
found++;
|
||||
/* We have a match -- play a greeting if they have it */
|
||||
res = play_mailbox_owner(chan, context, dialcontext, cat, name, readext, 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;
|
||||
res = 0;
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
free(conv);
|
||||
break;
|
||||
}
|
||||
free(conv);
|
||||
} else if (res == '*') {
|
||||
res = 0;
|
||||
v = v->next;
|
||||
} else {
|
||||
res = 0;
|
||||
goto ahem;
|
||||
}
|
||||
}
|
||||
} 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') {
|
||||
res = ast_streamfile(chan, found ? "dir-nomore" : "dir-nomatch", chan->language);
|
||||
if (!res)
|
||||
res = 1;
|
||||
return res;
|
||||
}
|
||||
return 0;
|
||||
|
||||
}
|
||||
return res;
|
||||
}
|
||||
@@ -566,113 +215,65 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, struct
|
||||
static int directory_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *u;
|
||||
struct ast_config *cfg, *ucfg;
|
||||
int last = 1;
|
||||
int readext = 0;
|
||||
int fromappvm = 0;
|
||||
const char *dirintro;
|
||||
char *parse;
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(vmcontext);
|
||||
AST_APP_ARG(dialcontext);
|
||||
AST_APP_ARG(options);
|
||||
);
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "Directory requires an argument (context[,dialcontext])\n");
|
||||
struct localuser *u;
|
||||
struct ast_config *cfg;
|
||||
if (!data) {
|
||||
ast_log(LOG_WARNING, "directory requires an argument (context)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
parse = ast_strdupa(data);
|
||||
|
||||
AST_STANDARD_APP_ARGS(args, parse);
|
||||
|
||||
if (args.options) {
|
||||
if (strchr(args.options, 'f'))
|
||||
last = 0;
|
||||
if (strchr(args.options, 'e'))
|
||||
readext = 1;
|
||||
if (strchr(args.options, 'v'))
|
||||
fromappvm = 1;
|
||||
}
|
||||
|
||||
if (ast_strlen_zero(args.dialcontext))
|
||||
args.dialcontext = args.vmcontext;
|
||||
|
||||
cfg = realtime_directory(args.vmcontext);
|
||||
cfg = ast_load(DIRECTORY_CONFIG);
|
||||
if (!cfg) {
|
||||
ast_log(LOG_ERROR, "Unable to read the configuration data!\n");
|
||||
ast_module_user_remove(u);
|
||||
ast_log(LOG_WARNING, "Unable to open directory configuration %s\n", DIRECTORY_CONFIG);
|
||||
return -1;
|
||||
}
|
||||
|
||||
ucfg = ast_config_load("users.conf");
|
||||
|
||||
dirintro = ast_variable_retrieve(cfg, args.vmcontext, "directoryintro");
|
||||
if (ast_strlen_zero(dirintro))
|
||||
dirintro = ast_variable_retrieve(cfg, "general", "directoryintro");
|
||||
if (ast_strlen_zero(dirintro))
|
||||
dirintro = last ? "dir-intro" : "dir-intro-fn";
|
||||
|
||||
if (chan->_state != AST_STATE_UP)
|
||||
res = ast_answer(chan);
|
||||
|
||||
for (;;) {
|
||||
if (!res)
|
||||
res = ast_stream_and_wait(chan, dirintro, chan->language, 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, ucfg, args.vmcontext, args.dialcontext, res, last, readext, 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;
|
||||
}
|
||||
if (ucfg)
|
||||
ast_config_destroy(ucfg);
|
||||
ast_config_destroy(cfg);
|
||||
ast_module_user_remove(u);
|
||||
ast_destroy(cfg);
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
res = ast_unregister_application(app);
|
||||
return res;
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
int load_module(void)
|
||||
{
|
||||
#ifdef ODBC_STORAGE
|
||||
struct ast_config *cfg = ast_config_load(VOICEMAIL_CONFIG);
|
||||
const 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);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Extension Directory");
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
|
||||
char *key()
|
||||
{
|
||||
return ASTERISK_GPL_KEY;
|
||||
}
|
||||
|
||||
617
apps/app_disa.c
617
apps/app_disa.c
@@ -1,389 +1,406 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
* Asterisk -- A telephony toolkit for Linux.
|
||||
*
|
||||
* Copyright (C) 1999 - 2005, Digium, Inc.
|
||||
* DISA -- Direct Inward System Access Application 6/20/2001
|
||||
*
|
||||
* Copyright (C) 2001, Linux Support Services, Inc.
|
||||
*
|
||||
*
|
||||
* 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.
|
||||
* Jim Dixon <jim@lambdatel.com>
|
||||
*
|
||||
* 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
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
#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/translate.h>
|
||||
#include <asterisk/ulaw.h>
|
||||
#include <string.h>
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <math.h>
|
||||
#include <pthread.h>
|
||||
#include <sys/time.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/indications.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/translate.h"
|
||||
#include "asterisk/ulaw.h"
|
||||
#include "asterisk/callerid.h"
|
||||
#include "asterisk/stringfields.h"
|
||||
#define TONE_BLOCK_SIZE 200
|
||||
|
||||
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"
|
||||
"DISA (Direct Inward System Access) -- Allows someone from outside\n"
|
||||
"the telephone switch (PBX) to obtain an \"internal\" system dialtone\n"
|
||||
"and to place calls from it as if they were placing a call from within\n"
|
||||
"the switch. A user calls a number that connects to the DISA application\n"
|
||||
"and is given dialtone. The user enters their passcode, followed by the\n"
|
||||
"pound sign (#). If the passcode is correct, the user is then given\n"
|
||||
"system dialtone on which a call may be placed. Obviously, this type\n"
|
||||
"of access has SERIOUS security implications, and GREAT care must be\n"
|
||||
"taken NOT to compromise your security.\n\n"
|
||||
"There is a possibility of accessing DISA without password. Simply\n"
|
||||
"exchange your password with \"no-password\".\n\n"
|
||||
" Example: exten => s,1,DISA(no-password|local)\n\n"
|
||||
"Be aware that using this compromises the security of your PBX.\n\n"
|
||||
"exchange your password with no-password.\n\n"
|
||||
" Example: exten => s,1,DISA,no-password|local\n\n"
|
||||
"but be aware of using this for your security compromising.\n\n"
|
||||
"The arguments to this application (in extensions.conf) allow either\n"
|
||||
"specification of a single global passcode (that everyone uses), or\n"
|
||||
"individual passcodes contained in a file. It also allows specification\n"
|
||||
"specification of a single global password (that everyone uses), or\n"
|
||||
"individual passwords contained in a file. It also allow specification\n"
|
||||
"of the context on which the user will be dialing. If no context is\n"
|
||||
"specified, the DISA application defaults the context to \"disa\".\n"
|
||||
"Presumably a normal system will have a special context set up\n"
|
||||
"for DISA use with some or a lot of restrictions. \n\n"
|
||||
"specified, the DISA application defaults the context to \"disa\"\n"
|
||||
"presumably that a normal system will have a special context set up\n"
|
||||
"for DISA use with some or a lot of restrictions. The arguments are\n"
|
||||
"one of the following:\n\n"
|
||||
" numeric-passcode\n"
|
||||
" numeric-passcode|context\n"
|
||||
" full-pathname-of-file-that-contains-passcodes\n\n"
|
||||
"The file that contains the passcodes (if used) allows specification\n"
|
||||
"of either just a passcode (defaulting to the \"disa\" context, or\n"
|
||||
"passcode|context on each line of the file. The file may contain blank\n"
|
||||
"lines, or comments starting with \"#\" or \";\". In addition, the\n"
|
||||
"above arguments may have |new-callerid-string appended to them, to\n"
|
||||
"specify a new (different) callerid to be used for this call, for\n"
|
||||
"example: numeric-passcode|context|\"My Phone\" <(234) 123-4567> or \n"
|
||||
"full-pathname-of-passcode-file|\"My Phone\" <(234) 123-4567>. 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. Also, if you set the 5th argument\n"
|
||||
"to 'NOANSWER', the DISA application will not answer initially.\n";
|
||||
"lines, or comments starting with \"#\" or \";\".\n\n"
|
||||
"If login is successful, the application parses the dialed number in\n"
|
||||
"the specified (or default) context, and returns 0 with the new extension\n"
|
||||
"context filled-in and the priority set to 1, so that the PBX may\n"
|
||||
"re-apply the routing tables to it and complete the call normally.";
|
||||
|
||||
|
||||
static void play_dialtone(struct ast_channel *chan, char *mailbox)
|
||||
STANDARD_LOCAL_USER;
|
||||
|
||||
LOCAL_USER_DECL;
|
||||
|
||||
static float loudness=8192.0;
|
||||
|
||||
int firstdigittimeout = 10000; /* 10 seconds first digit timeout */
|
||||
int digittimeout = 5000; /* 5 seconds subsequent digit timeout */
|
||||
|
||||
static void make_tone_block(unsigned char *data, float f1, float f2, int *x);
|
||||
|
||||
static void make_tone_block(unsigned char *data, float f1, float f2, int *x)
|
||||
{
|
||||
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);
|
||||
int i;
|
||||
float val;
|
||||
|
||||
for(i = 0; i < TONE_BLOCK_SIZE; i++)
|
||||
{
|
||||
val = loudness * sin((f1 * 2.0 * M_PI * (*x))/8000.0);
|
||||
val += loudness * sin((f2 * 2.0 * M_PI * (*x)++)/8000.0);
|
||||
data[i] = AST_LIN2MU((int)val);
|
||||
}
|
||||
/* wrap back around from 8000 */
|
||||
if (*x >= 8000) *x = 0;
|
||||
return;
|
||||
}
|
||||
|
||||
static int ms_diff(struct timeval *tv1, struct timeval *tv2)
|
||||
{
|
||||
int ms;
|
||||
|
||||
ms = (tv1->tv_sec - tv2->tv_sec) * 1000;
|
||||
ms += (tv1->tv_usec - tv2->tv_usec) / 1000;
|
||||
return(ms);
|
||||
}
|
||||
|
||||
static int disa_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int i,j,k,x,did_ignore,special_noanswer;
|
||||
int firstdigittimeout = 20000;
|
||||
int digittimeout = 10000;
|
||||
struct ast_module_user *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;
|
||||
int i,j,k,x;
|
||||
struct localuser *u;
|
||||
char tmp[256],exten[AST_MAX_EXTENSION],acctcode[20];
|
||||
unsigned char tone_block[TONE_BLOCK_SIZE],sil_block[TONE_BLOCK_SIZE];
|
||||
char *ourcontext;
|
||||
struct ast_frame *f,wf;
|
||||
fd_set readfds;
|
||||
int waitfor_notime;
|
||||
struct timeval notime = { 0,0 }, lastout, now, lastdigittime;
|
||||
FILE *fp;
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(passcode);
|
||||
AST_APP_ARG(context);
|
||||
AST_APP_ARG(cid);
|
||||
AST_APP_ARG(mailbox);
|
||||
AST_APP_ARG(noanswer);
|
||||
);
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "DISA requires an argument (passcode/passcode file)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
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);
|
||||
ast_module_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);
|
||||
ast_module_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);
|
||||
|
||||
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);
|
||||
|
||||
|
||||
special_noanswer = 0;
|
||||
if ((!args.noanswer) || strcmp(args.noanswer,"NOANSWER"))
|
||||
if (ast_set_write_format(chan,AST_FORMAT_ULAW))
|
||||
{
|
||||
if (chan->_state != AST_STATE_UP) {
|
||||
/* answer */
|
||||
ast_answer(chan);
|
||||
}
|
||||
} else special_noanswer = 1;
|
||||
ast_log(LOG_WARNING, "Unable to set write format to Mu-law on %s\n",chan->name);
|
||||
return -1;
|
||||
}
|
||||
if (ast_set_read_format(chan,AST_FORMAT_ULAW))
|
||||
{
|
||||
ast_log(LOG_WARNING, "Unable to set read format to Mu-law on %s\n",chan->name);
|
||||
return -1;
|
||||
}
|
||||
lastout.tv_sec = lastout.tv_usec = 0;
|
||||
/* make block of silence */
|
||||
memset(sil_block,0x7f,TONE_BLOCK_SIZE);
|
||||
if (!data || !strlen((char *)data)) {
|
||||
ast_log(LOG_WARNING, "disa requires an argument (passcode/passcode file)\n");
|
||||
return -1;
|
||||
}
|
||||
strncpy(tmp, (char *)data, sizeof(tmp)-1);
|
||||
strtok(tmp, "|");
|
||||
ourcontext = strtok(NULL, "|");
|
||||
/* if context not specified, use "disa" */
|
||||
if (!ourcontext) ourcontext = "disa";
|
||||
LOCAL_USER_ADD(u);
|
||||
if (chan->state != AST_STATE_UP)
|
||||
{
|
||||
/* answer */
|
||||
ast_answer(chan);
|
||||
}
|
||||
i = k = x = 0; /* k is 0 for pswd entry, 1 for ext entry */
|
||||
did_ignore = 0;
|
||||
exten[0] = 0;
|
||||
acctcode[0] = 0;
|
||||
/* can we access DISA without password? */
|
||||
|
||||
ast_log(LOG_DEBUG, "Context: %s\n",args.context);
|
||||
|
||||
if (!strcasecmp(args.passcode, "no-password")) {
|
||||
k |= 1; /* We have the password */
|
||||
if (!strcasecmp(tmp, "no-password"))
|
||||
{
|
||||
k = 1;
|
||||
ast_log(LOG_DEBUG, "DISA no-password login success\n");
|
||||
}
|
||||
lastdigittime = ast_tvnow();
|
||||
|
||||
play_dialtone(chan, args.mailbox);
|
||||
|
||||
for (;;) {
|
||||
gettimeofday(&lastdigittime,NULL);
|
||||
for(;;)
|
||||
{
|
||||
gettimeofday(&now,NULL);
|
||||
/* if outa time, give em reorder */
|
||||
if (ast_tvdiff_ms(ast_tvnow(), lastdigittime) >
|
||||
((k&2) ? digittimeout : firstdigittimeout)) {
|
||||
if (ms_diff(&now,&lastdigittime) >
|
||||
((k) ? digittimeout : firstdigittimeout))
|
||||
{
|
||||
ast_log(LOG_DEBUG,"DISA %s entry timeout on chan %s\n",
|
||||
((k&1) ? "extension" : "password"),chan->name);
|
||||
break;
|
||||
((k) ? "extension" : "password"),chan->name);
|
||||
goto reorder;
|
||||
}
|
||||
if ((res = ast_waitfor(chan, -1) < 0)) {
|
||||
ast_log(LOG_DEBUG, "Waitfor returned %d\n", res);
|
||||
continue;
|
||||
/* if first digit or ignore, send dialtone */
|
||||
if ((!i) || (ast_ignore_pattern(ourcontext,exten) && k))
|
||||
{
|
||||
gettimeofday(&now,NULL);
|
||||
if (lastout.tv_sec &&
|
||||
(ms_diff(&now,&lastout) < 25)) continue;
|
||||
lastout.tv_sec = now.tv_sec;
|
||||
lastout.tv_usec = now.tv_usec;
|
||||
wf.frametype = AST_FRAME_VOICE;
|
||||
wf.subclass = AST_FORMAT_ULAW;
|
||||
wf.offset = AST_FRIENDLY_OFFSET;
|
||||
wf.mallocd = 0;
|
||||
wf.data = tone_block;
|
||||
wf.datalen = TONE_BLOCK_SIZE;
|
||||
/* make this tone block */
|
||||
make_tone_block(tone_block,350.0,440.0,&x);
|
||||
wf.timelen = wf.datalen / 8;
|
||||
if (ast_write(chan, &wf))
|
||||
{
|
||||
ast_log(LOG_WARNING, "DISA Failed to write frame on %s\n",chan->name);
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
|
||||
waitfor_notime = notime.tv_usec + notime.tv_sec * 1000;
|
||||
if (!ast_waitfor_nandfds(&chan, 1, &(chan->fds[0]), 1, NULL, NULL,
|
||||
&waitfor_notime)) continue;
|
||||
f = ast_read(chan);
|
||||
if (f == NULL) {
|
||||
ast_module_user_remove(u);
|
||||
if (f == NULL)
|
||||
{
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return -1;
|
||||
}
|
||||
if ((f->frametype == AST_FRAME_CONTROL) &&
|
||||
(f->subclass == AST_CONTROL_HANGUP)) {
|
||||
(f->subclass == AST_CONTROL_HANGUP))
|
||||
{
|
||||
ast_frfree(f);
|
||||
ast_module_user_remove(u);
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return -1;
|
||||
}
|
||||
if (f->frametype == AST_FRAME_VOICE) {
|
||||
/* if not DTMF, just do it again */
|
||||
if (f->frametype != AST_FRAME_DTMF)
|
||||
{
|
||||
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();
|
||||
gettimeofday(&lastdigittime,NULL);
|
||||
/* 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 */
|
||||
if (i < AST_MAX_EXTENSION) /* if still valid number of digits */
|
||||
{
|
||||
if (!k) /* if in password state */
|
||||
{
|
||||
if (j == '#') /* end of password */
|
||||
{
|
||||
/* see if this is an integer */
|
||||
if (sscanf(args.passcode,"%d",&j) < 1) { /* nope, it must be a filename */
|
||||
fp = fopen(args.passcode,"r");
|
||||
if (!fp) {
|
||||
ast_log(LOG_WARNING,"DISA password file %s not found on chan %s\n",args.passcode,chan->name);
|
||||
ast_module_user_remove(u);
|
||||
if (sscanf(tmp,"%d",&j) < 1)
|
||||
{ /* nope, it must be a filename */
|
||||
fp = fopen(tmp,"r");
|
||||
if (!fp)
|
||||
{
|
||||
ast_log(LOG_WARNING,"DISA password file %s not found on chan %s\n",tmp,chan->name);
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return -1;
|
||||
}
|
||||
pwline[0] = 0;
|
||||
while(fgets(pwline,sizeof(pwline) - 1,fp)) {
|
||||
if (!pwline[0])
|
||||
continue;
|
||||
if (pwline[strlen(pwline) - 1] == '\n')
|
||||
pwline[strlen(pwline) - 1] = 0;
|
||||
if (!pwline[0])
|
||||
continue;
|
||||
/* skip comments */
|
||||
if (pwline[0] == '#')
|
||||
continue;
|
||||
if (pwline[0] == ';')
|
||||
continue;
|
||||
|
||||
AST_STANDARD_APP_ARGS(args, pwline);
|
||||
|
||||
ast_log(LOG_DEBUG, "Mailbox: %s\n",args.mailbox);
|
||||
|
||||
/* password must be in valid format (numeric) */
|
||||
if (sscanf(args.passcode,"%d", &j) < 1)
|
||||
continue;
|
||||
/* if we got it */
|
||||
if (!strcmp(exten,args.passcode)) {
|
||||
if (ast_strlen_zero(args.context))
|
||||
args.context = "disa";
|
||||
if (ast_strlen_zero(args.mailbox))
|
||||
args.mailbox = "";
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
tmp[0] = 0;
|
||||
while(fgets(tmp,sizeof(tmp) - 1,fp))
|
||||
{
|
||||
if (!tmp[0]) continue;
|
||||
if (tmp[strlen(tmp) - 1] == '\n')
|
||||
tmp[strlen(tmp) - 1] = 0;
|
||||
if (!tmp[0]) continue;
|
||||
/* skip comments */
|
||||
if (tmp[0] == '#') continue;
|
||||
if (tmp[0] == ';') continue;
|
||||
strtok(tmp, "|");
|
||||
ourcontext = strtok(NULL, "|");
|
||||
/* password must be in valid format (numeric) */
|
||||
if (sscanf(tmp,"%d",&j) < 1) continue;
|
||||
/* if we got it */
|
||||
if (!strcmp(exten,tmp)) break;
|
||||
}
|
||||
fclose(fp);
|
||||
}
|
||||
/* compare the two */
|
||||
if (strcmp(exten,args.passcode)) {
|
||||
}
|
||||
/* compare the two */
|
||||
if (strcmp(exten,tmp))
|
||||
{
|
||||
ast_log(LOG_WARNING,"DISA on chan %s got bad password %s\n",chan->name,exten);
|
||||
goto reorder;
|
||||
|
||||
}
|
||||
/* password good, set to dial state */
|
||||
ast_log(LOG_DEBUG,"DISA on chan %s password is good\n",chan->name);
|
||||
play_dialtone(chan, args.mailbox);
|
||||
|
||||
k|=1; /* In number mode */
|
||||
k = 1;
|
||||
i = 0; /* re-set buffer pointer */
|
||||
exten[sizeof(acctcode)] = 0;
|
||||
ast_copy_string(acctcode, exten, sizeof(acctcode));
|
||||
strcpy(acctcode,exten);
|
||||
exten[0] = 0;
|
||||
ast_log(LOG_DEBUG,"Successful DISA log-in on chan %s\n", chan->name);
|
||||
ast_log(LOG_DEBUG,"Successful DISA log-in on chan %s\n",chan->name);
|
||||
continue;
|
||||
}
|
||||
}
|
||||
|
||||
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) continue; /* if getting password, continue doing it */
|
||||
/* if this exists */
|
||||
if (ast_exists_extension(chan,ourcontext,exten,1, chan->callerid))
|
||||
{
|
||||
strcpy(chan->exten,exten);
|
||||
strcpy(chan->context,ourcontext);
|
||||
strcpy(chan->accountcode,acctcode);
|
||||
chan->priority = 0;
|
||||
ast_cdr_init(chan->cdr,chan);
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return 0;
|
||||
}
|
||||
/* if can do some more, do it */
|
||||
if (ast_canmatch_extension(chan,ourcontext,exten,1, chan->callerid)) continue;
|
||||
}
|
||||
}
|
||||
|
||||
if (k == 3) {
|
||||
int recheck = 0;
|
||||
struct ast_flags flags = { AST_CDR_FLAG_POSTED };
|
||||
|
||||
if (!ast_exists_extension(chan, args.context, exten, 1, chan->cid.cid_num)) {
|
||||
pbx_builtin_setvar_helper(chan, "INVALID_EXTEN", exten);
|
||||
exten[0] = 'i';
|
||||
exten[1] = '\0';
|
||||
recheck = 1;
|
||||
}
|
||||
if (!recheck || ast_exists_extension(chan, args.context, exten, 1, chan->cid.cid_num)) {
|
||||
ast_playtones_stop(chan);
|
||||
/* We're authenticated and have a target extension */
|
||||
if (!ast_strlen_zero(args.cid)) {
|
||||
ast_callerid_split(args.cid, ourcidname, sizeof(ourcidname), ourcidnum, sizeof(ourcidnum));
|
||||
ast_set_callerid(chan, ourcidnum, ourcidname, ourcidnum);
|
||||
}
|
||||
|
||||
if (!ast_strlen_zero(acctcode))
|
||||
ast_string_field_set(chan, accountcode, acctcode);
|
||||
|
||||
if (special_noanswer) flags.flags = 0;
|
||||
ast_cdr_reset(chan->cdr, &flags);
|
||||
ast_explicit_goto(chan, args.context, exten, 1);
|
||||
ast_module_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);
|
||||
/* something is invalid, give em reorder forever */
|
||||
x = 0;
|
||||
for(;;)
|
||||
{
|
||||
for(i = 0; i < 10; i++)
|
||||
{
|
||||
do gettimeofday(&now,NULL);
|
||||
while (lastout.tv_sec &&
|
||||
(ms_diff(&now,&lastout) < 25)) ;
|
||||
lastout.tv_sec = now.tv_sec;
|
||||
lastout.tv_usec = now.tv_usec;
|
||||
wf.frametype = AST_FRAME_VOICE;
|
||||
wf.subclass = AST_FORMAT_ULAW;
|
||||
wf.offset = AST_FRIENDLY_OFFSET;
|
||||
wf.mallocd = 0;
|
||||
wf.data = tone_block;
|
||||
wf.datalen = TONE_BLOCK_SIZE;
|
||||
/* make this tone block */
|
||||
make_tone_block(tone_block,480.0,620.0,&x);
|
||||
wf.timelen = wf.datalen / 8;
|
||||
if (ast_write(chan, &wf))
|
||||
{
|
||||
ast_log(LOG_WARNING, "DISA Failed to write frame on %s\n",chan->name);
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return -1;
|
||||
}
|
||||
FD_ZERO(&readfds);
|
||||
FD_SET(chan->fds[0],&readfds);
|
||||
/* if no read avail, do send again */
|
||||
if (select(chan->fds[0] + 1,&readfds,NULL,
|
||||
NULL,¬ime) < 1) continue;
|
||||
/* read frame */
|
||||
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;
|
||||
}
|
||||
ast_frfree(f);
|
||||
}
|
||||
for(i = 0; i < 10; i++)
|
||||
{
|
||||
do gettimeofday(&now,NULL);
|
||||
while (lastout.tv_sec &&
|
||||
(ms_diff(&now,&lastout) < 25)) ;
|
||||
lastout.tv_sec = now.tv_sec;
|
||||
lastout.tv_usec = now.tv_usec;
|
||||
wf.frametype = AST_FRAME_VOICE;
|
||||
wf.subclass = AST_FORMAT_ULAW;
|
||||
wf.offset = AST_FRIENDLY_OFFSET;
|
||||
wf.mallocd = 0;
|
||||
wf.data = sil_block;
|
||||
wf.datalen = TONE_BLOCK_SIZE;
|
||||
wf.timelen = wf.datalen / 8;
|
||||
if (ast_write(chan, &wf))
|
||||
{
|
||||
ast_log(LOG_WARNING, "DISA Failed to write frame on %s\n",chan->name);
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return -1;
|
||||
}
|
||||
FD_ZERO(&readfds);
|
||||
FD_SET(chan->fds[0],&readfds);
|
||||
/* if no read avail, do send again */
|
||||
if (select(chan->fds[0] + 1,&readfds,NULL,
|
||||
NULL,¬ime) < 1) continue;
|
||||
/* read frame */
|
||||
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;
|
||||
}
|
||||
ast_frfree(f);
|
||||
}
|
||||
}
|
||||
}
|
||||
ast_playtones_stop(chan);
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, disa_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "DISA (Direct Inward System Access) Application");
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
|
||||
char *key()
|
||||
{
|
||||
return ASTERISK_GPL_KEY;
|
||||
}
|
||||
|
||||
@@ -1,176 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 2004 - 2005, Anthony Minessale II.
|
||||
*
|
||||
* Anthony Minessale <anthmct@yahoo.com>
|
||||
*
|
||||
* A license has been granted to Digium (via disclaimer) for the use of
|
||||
* this code.
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief Application to dump channel variables
|
||||
*
|
||||
* \author Anthony Minessale <anthmct@yahoo.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/utils.h"
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/utils.h"
|
||||
|
||||
static char *app = "DumpChan";
|
||||
static char *synopsis = "Dump Info About The Calling Channel";
|
||||
static char *desc =
|
||||
" DumpChan([<min_verbose_level>])\n"
|
||||
"Displays information on channel and listing of all channel\n"
|
||||
"variables. If min_verbose_level is specified, output is only\n"
|
||||
"displayed when the verbose level is currently set to that number\n"
|
||||
"or greater. \n";
|
||||
|
||||
|
||||
static int serialize_showchan(struct ast_channel *c, char *buf, size_t size)
|
||||
{
|
||||
struct timeval now;
|
||||
long elapsed_seconds = 0;
|
||||
int hour = 0, min = 0, sec = 0;
|
||||
char cgrp[BUFSIZ/2];
|
||||
char pgrp[BUFSIZ/2];
|
||||
char formatbuf[BUFSIZ/2];
|
||||
|
||||
now = ast_tvnow();
|
||||
memset(buf, 0, size);
|
||||
if (!c)
|
||||
return 0;
|
||||
|
||||
if (c->cdr) {
|
||||
elapsed_seconds = now.tv_sec - c->cdr->start.tv_sec;
|
||||
hour = elapsed_seconds / 3600;
|
||||
min = (elapsed_seconds % 3600) / 60;
|
||||
sec = elapsed_seconds % 60;
|
||||
}
|
||||
|
||||
snprintf(buf,size,
|
||||
"Name= %s\n"
|
||||
"Type= %s\n"
|
||||
"UniqueID= %s\n"
|
||||
"CallerID= %s\n"
|
||||
"CallerIDName= %s\n"
|
||||
"DNIDDigits= %s\n"
|
||||
"RDNIS= %s\n"
|
||||
"State= %s (%d)\n"
|
||||
"Rings= %d\n"
|
||||
"NativeFormat= %s\n"
|
||||
"WriteFormat= %s\n"
|
||||
"ReadFormat= %s\n"
|
||||
"1stFileDescriptor= %d\n"
|
||||
"Framesin= %d %s\n"
|
||||
"Framesout= %d %s\n"
|
||||
"TimetoHangup= %ld\n"
|
||||
"ElapsedTime= %dh%dm%ds\n"
|
||||
"Context= %s\n"
|
||||
"Extension= %s\n"
|
||||
"Priority= %d\n"
|
||||
"CallGroup= %s\n"
|
||||
"PickupGroup= %s\n"
|
||||
"Application= %s\n"
|
||||
"Data= %s\n"
|
||||
"Blocking_in= %s\n",
|
||||
c->name,
|
||||
c->tech->type,
|
||||
c->uniqueid,
|
||||
S_OR(c->cid.cid_num, "(N/A)"),
|
||||
S_OR(c->cid.cid_name, "(N/A)"),
|
||||
S_OR(c->cid.cid_dnid, "(N/A)"),
|
||||
S_OR(c->cid.cid_rdnis, "(N/A)"),
|
||||
ast_state2str(c->_state),
|
||||
c->_state,
|
||||
c->rings,
|
||||
ast_getformatname_multiple(formatbuf, sizeof(formatbuf), c->nativeformats),
|
||||
ast_getformatname_multiple(formatbuf, sizeof(formatbuf), c->writeformat),
|
||||
ast_getformatname_multiple(formatbuf, sizeof(formatbuf), c->readformat),
|
||||
c->fds[0], c->fin & ~DEBUGCHAN_FLAG, (c->fin & DEBUGCHAN_FLAG) ? " (DEBUGGED)" : "",
|
||||
c->fout & ~DEBUGCHAN_FLAG, (c->fout & DEBUGCHAN_FLAG) ? " (DEBUGGED)" : "", (long)c->whentohangup,
|
||||
hour,
|
||||
min,
|
||||
sec,
|
||||
c->context,
|
||||
c->exten,
|
||||
c->priority,
|
||||
ast_print_group(cgrp, sizeof(cgrp), c->callgroup),
|
||||
ast_print_group(pgrp, sizeof(pgrp), c->pickupgroup),
|
||||
( c->appl ? c->appl : "(N/A)" ),
|
||||
( c-> data ? S_OR(c->data, "(Empty)") : "(None)"),
|
||||
(ast_test_flag(c, AST_FLAG_BLOCKING) ? c->blockproc : "(Not Blocking)"));
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int dumpchan_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *u;
|
||||
char vars[BUFSIZ * 4];
|
||||
char info[1024];
|
||||
int level = 0;
|
||||
static char *line = "================================================================================";
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (!ast_strlen_zero(data))
|
||||
level = atoi(data);
|
||||
|
||||
pbx_builtin_serialize_variables(chan, vars, sizeof(vars));
|
||||
serialize_showchan(chan, info, sizeof(info));
|
||||
if (option_verbose >= level)
|
||||
ast_verbose("\nDumping Info For Channel: %s:\n%s\nInfo:\n%s\nVariables:\n%s%s\n", chan->name, line, info, vars, line);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, dumpchan_exec, synopsis, desc);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Dump Info About The Calling Channel");
|
||||
132
apps/app_echo.c
132
apps/app_echo.c
@@ -1,104 +1,94 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
* Asterisk -- A telephony toolkit for Linux.
|
||||
*
|
||||
* Copyright (C) 1999 - 2005, Digium, Inc.
|
||||
* Echo application -- play back what you hear to evaluate latency
|
||||
*
|
||||
* Copyright (C) 1999, Mark Spencer
|
||||
*
|
||||
* Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
* Mark Spencer <markster@linux-support.net>
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief Echo application -- play back what you hear to evaluate latency
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.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 <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
#include <stdlib.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include <pthread.h>
|
||||
|
||||
|
||||
static char *tdesc = "Simple Echo Application";
|
||||
|
||||
static char *app = "Echo";
|
||||
|
||||
static char *synopsis = "Echo audio, video, or DTMF back to the calling party";
|
||||
static char *synopsis = "Echo audio read back to the user";
|
||||
|
||||
static char *descrip =
|
||||
" Echo(): This application will echo any audio, video, or DTMF frames read from\n"
|
||||
"the calling channel back to itself. If the DTMF digit '#' is received, the\n"
|
||||
"application will exit.\n";
|
||||
" Echo(): Echo audio read from channel back to the channel. Returns 0\n"
|
||||
"if the user exits with the '#' key, or -1 if the user hangs up.\n";
|
||||
|
||||
STANDARD_LOCAL_USER;
|
||||
|
||||
LOCAL_USER_DECL;
|
||||
|
||||
static int echo_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = -1;
|
||||
int format;
|
||||
struct ast_module_user *u;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
format = ast_best_codec(chan->nativeformats);
|
||||
ast_set_write_format(chan, format);
|
||||
ast_set_read_format(chan, format);
|
||||
|
||||
while (ast_waitfor(chan, -1) > -1) {
|
||||
struct ast_frame *f = ast_read(chan);
|
||||
if (!f)
|
||||
break;
|
||||
f->delivery.tv_sec = 0;
|
||||
f->delivery.tv_usec = 0;
|
||||
if (ast_write(chan, f)) {
|
||||
ast_frfree(f);
|
||||
goto end;
|
||||
}
|
||||
if ((f->frametype == AST_FRAME_DTMF) && (f->subclass == '#')) {
|
||||
res = 0;
|
||||
ast_frfree(f);
|
||||
goto end;
|
||||
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((f = ast_read(chan))) {
|
||||
if (f->frametype == AST_FRAME_VOICE) {
|
||||
if (ast_write(chan, f))
|
||||
break;
|
||||
} else if (f->frametype == AST_FRAME_DTMF) {
|
||||
if (f->subclass == '#') {
|
||||
res = 0;
|
||||
break;
|
||||
} else
|
||||
if (ast_write(chan, f))
|
||||
break;
|
||||
}
|
||||
ast_frfree(f);
|
||||
}
|
||||
end:
|
||||
ast_module_user_remove(u);
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, echo_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Simple Echo Application");
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
|
||||
char *key()
|
||||
{
|
||||
return ASTERISK_GPL_KEY;
|
||||
}
|
||||
|
||||
221
apps/app_exec.c
221
apps/app_exec.c
@@ -1,221 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (c) 2004 - 2005, Tilghman Lesher. All rights reserved.
|
||||
* Portions copyright (c) 2006, Philipp Dunkel.
|
||||
*
|
||||
* Tilghman Lesher <app_exec__v002@the-tilghman.com>
|
||||
*
|
||||
* This code is released by the author with no restrictions on usage.
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief Exec application
|
||||
*
|
||||
* \author Tilghman Lesher <app_exec__v002@the-tilghman.com>
|
||||
* \author Philipp Dunkel <philipp.dunkel@ebox.at>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
|
||||
/* Maximum length of any variable */
|
||||
#define MAXRESULT 1024
|
||||
|
||||
/*! Note
|
||||
*
|
||||
* The key difference between these two apps is exit status. In a
|
||||
* nutshell, Exec tries to be transparent as possible, behaving
|
||||
* in exactly the same way as if the application it calls was
|
||||
* directly invoked from the dialplan.
|
||||
*
|
||||
* TryExec, on the other hand, provides a way to execute applications
|
||||
* and catch any possible fatal error without actually fatally
|
||||
* affecting the dialplan.
|
||||
*/
|
||||
|
||||
static char *app_exec = "Exec";
|
||||
static char *exec_synopsis = "Executes dialplan 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. If the underlying application\n"
|
||||
"terminates the dialplan, or if the application cannot be found,\n"
|
||||
"Exec will terminate the dialplan.\n"
|
||||
" To invoke external applications, see the application System.\n"
|
||||
" If you would like to catch any error instead, see TryExec.\n";
|
||||
|
||||
static char *app_tryexec = "TryExec";
|
||||
static char *tryexec_synopsis = "Executes dialplan application, always returning";
|
||||
static char *tryexec_descrip =
|
||||
"Usage: TryExec(appname(arguments))\n"
|
||||
" Allows an arbitrary application to be invoked even when not\n"
|
||||
"hardcoded into the dialplan. To invoke external applications\n"
|
||||
"see the application System. Always returns to the dialplan.\n"
|
||||
"The channel variable TRYSTATUS will be set to:\n"
|
||||
" SUCCESS if the application returned zero\n"
|
||||
" FAILED if the application returned non-zero\n"
|
||||
" NOAPP if the application was not found or was not specified\n";
|
||||
|
||||
static char *app_execif = "ExecIf";
|
||||
static char *execif_synopsis = "Executes dialplan application, conditionally";
|
||||
static char *execif_descrip =
|
||||
"Usage: ExecIF (<expr>|<app>|<data>)\n"
|
||||
"If <expr> is true, execute and return the result of <app>(<data>).\n"
|
||||
"If <expr> is true, but <app> is not found, then the application\n"
|
||||
"will return a non-zero value.\n";
|
||||
|
||||
static int exec_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res=0;
|
||||
struct ast_module_user *u;
|
||||
char *s, *appname, *endargs, args[MAXRESULT] = "";
|
||||
struct ast_app *app;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
/* Check and parse arguments */
|
||||
if (data) {
|
||||
s = ast_strdupa(data);
|
||||
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);
|
||||
} else {
|
||||
ast_log(LOG_WARNING, "Could not find application (%s)\n", appname);
|
||||
res = -1;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int tryexec_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res=0;
|
||||
struct ast_module_user *u;
|
||||
char *s, *appname, *endargs, args[MAXRESULT] = "";
|
||||
struct ast_app *app;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
/* Check and parse arguments */
|
||||
if (data) {
|
||||
s = ast_strdupa(data);
|
||||
appname = strsep(&s, "(");
|
||||
if (s) {
|
||||
endargs = strrchr(s, ')');
|
||||
if (endargs)
|
||||
*endargs = '\0';
|
||||
pbx_substitute_variables_helper(chan, s, args, MAXRESULT - 1);
|
||||
}
|
||||
if (appname) {
|
||||
app = pbx_findapp(appname);
|
||||
if (app) {
|
||||
res = pbx_exec(chan, app, args);
|
||||
pbx_builtin_setvar_helper(chan, "TRYSTATUS", res ? "FAILED" : "SUCCESS");
|
||||
} else {
|
||||
ast_log(LOG_WARNING, "Could not find application (%s)\n", appname);
|
||||
pbx_builtin_setvar_helper(chan, "TRYSTATUS", "NOAPP");
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int execif_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *u;
|
||||
char *myapp = NULL;
|
||||
char *mydata = NULL;
|
||||
char *expr = NULL;
|
||||
struct ast_app *app = NULL;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
expr = ast_strdupa(data);
|
||||
|
||||
if ((myapp = strchr(expr,'|'))) {
|
||||
*myapp = '\0';
|
||||
myapp++;
|
||||
if ((mydata = strchr(myapp,'|'))) {
|
||||
*mydata = '\0';
|
||||
mydata++;
|
||||
} else
|
||||
mydata = "";
|
||||
|
||||
if (pbx_checkcondition(expr)) {
|
||||
if ((app = pbx_findapp(myapp))) {
|
||||
res = pbx_exec(chan, app, mydata);
|
||||
} else {
|
||||
ast_log(LOG_WARNING, "Count not find application! (%s)\n", myapp);
|
||||
res = -1;
|
||||
}
|
||||
}
|
||||
} else {
|
||||
ast_log(LOG_ERROR,"Invalid Syntax.\n");
|
||||
res = -1;
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app_exec);
|
||||
res |= ast_unregister_application(app_tryexec);
|
||||
res |= ast_unregister_application(app_execif);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
int res = ast_register_application(app_exec, exec_exec, exec_synopsis, exec_descrip);
|
||||
res |= ast_register_application(app_tryexec, tryexec_exec, tryexec_synopsis, tryexec_descrip);
|
||||
res |= ast_register_application(app_execif, execif_exec, execif_synopsis, execif_descrip);
|
||||
return res;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Executes dialplan applications");
|
||||
@@ -1,578 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 1999 - 2005, Digium, Inc.
|
||||
*
|
||||
* Kevin P. Fleming <kpfleming@digium.com>
|
||||
*
|
||||
* Portions taken from the file-based music-on-hold work
|
||||
* created by Anthony Minessale II in res_musiconhold.c
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief External IVR application interface
|
||||
*
|
||||
* \author Kevin P. Fleming <kpfleming@digium.com>
|
||||
*
|
||||
* \note Portions taken from the file-based music-on-hold work
|
||||
* created by Anthony Minessale II in res_musiconhold.c
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
#include <errno.h>
|
||||
#include <signal.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/linkedlists.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/utils.h"
|
||||
#include "asterisk/options.h"
|
||||
|
||||
static const char *app = "ExternalIVR";
|
||||
|
||||
static const char *synopsis = "Interfaces with an external IVR application";
|
||||
|
||||
static const char *descrip =
|
||||
" ExternalIVR(command[|arg[|arg...]]): Forks an process to run the supplied command,\n"
|
||||
"and starts a generator on the channel. The generator's play list is\n"
|
||||
"controlled by the external application, which can add and clear entries\n"
|
||||
"via simple commands issued over its stdout. The external application\n"
|
||||
"will receive all DTMF events received on the channel, and notification\n"
|
||||
"if the channel is hung up. The application will not be forcibly terminated\n"
|
||||
"when the channel is hung up.\n"
|
||||
"See doc/externalivr.txt for a protocol specification.\n";
|
||||
|
||||
/* XXX the parser in gcc 2.95 gets confused if you don't put a space between 'name' and the comma */
|
||||
#define ast_chan_log(level, channel, format, ...) ast_log(level, "%s: " format, channel->name , ## __VA_ARGS__)
|
||||
|
||||
struct playlist_entry {
|
||||
AST_LIST_ENTRY(playlist_entry) list;
|
||||
char filename[1];
|
||||
};
|
||||
|
||||
struct ivr_localuser {
|
||||
struct ast_channel *chan;
|
||||
AST_LIST_HEAD(playlist, playlist_entry) playlist;
|
||||
AST_LIST_HEAD(finishlist, playlist_entry) finishlist;
|
||||
int abort_current_sound;
|
||||
int playing_silence;
|
||||
int option_autoclear;
|
||||
};
|
||||
|
||||
|
||||
struct gen_state {
|
||||
struct ivr_localuser *u;
|
||||
struct ast_filestream *stream;
|
||||
struct playlist_entry *current;
|
||||
int sample_queue;
|
||||
};
|
||||
|
||||
static 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 ivr_localuser *u = params;
|
||||
struct gen_state *state;
|
||||
|
||||
if (!(state = ast_calloc(1, sizeof(*state))))
|
||||
return NULL;
|
||||
|
||||
state->u = u;
|
||||
|
||||
return state;
|
||||
}
|
||||
|
||||
static void gen_closestream(struct gen_state *state)
|
||||
{
|
||||
if (!state->stream)
|
||||
return;
|
||||
|
||||
ast_closestream(state->stream);
|
||||
state->u->chan->stream = NULL;
|
||||
state->stream = NULL;
|
||||
}
|
||||
|
||||
static void gen_release(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct gen_state *state = data;
|
||||
|
||||
gen_closestream(state);
|
||||
free(data);
|
||||
}
|
||||
|
||||
/* caller has the playlist locked */
|
||||
static int gen_nextfile(struct gen_state *state)
|
||||
{
|
||||
struct ivr_localuser *u = state->u;
|
||||
char *file_to_stream;
|
||||
|
||||
u->abort_current_sound = 0;
|
||||
u->playing_silence = 0;
|
||||
gen_closestream(state);
|
||||
|
||||
while (!state->stream) {
|
||||
state->current = AST_LIST_REMOVE_HEAD(&u->playlist, list);
|
||||
if (state->current) {
|
||||
file_to_stream = state->current->filename;
|
||||
} else {
|
||||
file_to_stream = "silence/10";
|
||||
u->playing_silence = 1;
|
||||
}
|
||||
|
||||
if (!(state->stream = ast_openstream_full(u->chan, file_to_stream, u->chan->language, 1))) {
|
||||
ast_chan_log(LOG_WARNING, u->chan, "File '%s' could not be opened: %s\n", file_to_stream, strerror(errno));
|
||||
if (!u->playing_silence) {
|
||||
continue;
|
||||
} else {
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
return (!state->stream);
|
||||
}
|
||||
|
||||
static struct ast_frame *gen_readframe(struct gen_state *state)
|
||||
{
|
||||
struct ast_frame *f = NULL;
|
||||
struct ivr_localuser *u = state->u;
|
||||
|
||||
if (u->abort_current_sound ||
|
||||
(u->playing_silence && AST_LIST_FIRST(&u->playlist))) {
|
||||
gen_closestream(state);
|
||||
AST_LIST_LOCK(&u->playlist);
|
||||
gen_nextfile(state);
|
||||
AST_LIST_UNLOCK(&u->playlist);
|
||||
}
|
||||
|
||||
if (!(state->stream && (f = ast_readframe(state->stream)))) {
|
||||
if (state->current) {
|
||||
AST_LIST_LOCK(&u->finishlist);
|
||||
AST_LIST_INSERT_TAIL(&u->finishlist, state->current, list);
|
||||
AST_LIST_UNLOCK(&u->finishlist);
|
||||
state->current = NULL;
|
||||
}
|
||||
if (!gen_nextfile(state))
|
||||
f = ast_readframe(state->stream);
|
||||
}
|
||||
|
||||
return f;
|
||||
}
|
||||
|
||||
static int gen_generate(struct ast_channel *chan, void *data, int len, int samples)
|
||||
{
|
||||
struct gen_state *state = data;
|
||||
struct ast_frame *f = NULL;
|
||||
int res = 0;
|
||||
|
||||
state->sample_queue += samples;
|
||||
|
||||
while (state->sample_queue > 0) {
|
||||
if (!(f = gen_readframe(state)))
|
||||
return -1;
|
||||
|
||||
res = ast_write(chan, f);
|
||||
ast_frfree(f);
|
||||
if (res < 0) {
|
||||
ast_chan_log(LOG_WARNING, chan, "Failed to write frame: %s\n", strerror(errno));
|
||||
return -1;
|
||||
}
|
||||
state->sample_queue -= f->samples;
|
||||
}
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static struct ast_generator gen =
|
||||
{
|
||||
alloc: gen_alloc,
|
||||
release: gen_release,
|
||||
generate: gen_generate,
|
||||
};
|
||||
|
||||
static struct playlist_entry *make_entry(const char *filename)
|
||||
{
|
||||
struct playlist_entry *entry;
|
||||
|
||||
if (!(entry = ast_calloc(1, sizeof(*entry) + strlen(filename) + 10))) /* XXX why 10 ? */
|
||||
return NULL;
|
||||
|
||||
strcpy(entry->filename, filename);
|
||||
|
||||
return entry;
|
||||
}
|
||||
|
||||
static int app_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *lu;
|
||||
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;
|
||||
struct ivr_localuser foo = {
|
||||
.playlist = AST_LIST_HEAD_INIT_VALUE,
|
||||
.finishlist = AST_LIST_HEAD_INIT_VALUE,
|
||||
};
|
||||
struct ivr_localuser *u = &foo;
|
||||
sigset_t fullset, oldset;
|
||||
|
||||
lu = ast_module_user_add(chan);
|
||||
|
||||
sigfillset(&fullset);
|
||||
pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
|
||||
|
||||
u->abort_current_sound = 0;
|
||||
u->chan = chan;
|
||||
|
||||
if (ast_strlen_zero(args)) {
|
||||
ast_log(LOG_WARNING, "ExternalIVR requires a command to execute\n");
|
||||
ast_module_user_remove(lu);
|
||||
return -1;
|
||||
}
|
||||
|
||||
buf = ast_strdupa(data);
|
||||
|
||||
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 (ast_opt_high_priority)
|
||||
ast_set_priority(0);
|
||||
|
||||
dup2(child_stdin[0], STDIN_FILENO);
|
||||
dup2(child_stdout[1], STDOUT_FILENO);
|
||||
dup2(child_stderr[1], STDERR_FILENO);
|
||||
for (i = STDERR_FILENO + 1; i < 1024; i++)
|
||||
close(i);
|
||||
execv(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);
|
||||
|
||||
ast_module_user_remove(lu);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, app_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "External IVR Interface Application");
|
||||
@@ -1,553 +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
|
||||
*
|
||||
* \author Christos Ricudis <ricudis@itc.auth.gr>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#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/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 *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";
|
||||
|
||||
|
||||
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;
|
||||
if (!(buff = ast_malloc(bufflen)))
|
||||
{
|
||||
/* TODO: Handle memory allocation failure */
|
||||
}
|
||||
*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;
|
||||
if (!(buff = ast_realloc(buff, bufflen)))
|
||||
{
|
||||
/* TODO: Handle memory allocation failure */
|
||||
}
|
||||
}
|
||||
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 (ast_opt_high_priority)
|
||||
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 = {
|
||||
.f = { 0, },
|
||||
};
|
||||
|
||||
if (pipe(fds)) {
|
||||
ast_log(LOG_WARNING, "Unable to create pipe\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* Answer if it's not already going */
|
||||
if (chan->_state != AST_STATE_UP)
|
||||
ast_answer(chan);
|
||||
ast_stopstream(chan);
|
||||
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.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 ast_module_user *u;
|
||||
struct sockaddr_in serv_addr;
|
||||
struct hostent *serverhost;
|
||||
struct ast_hostent ahp;
|
||||
int fd;
|
||||
FILE *fs;
|
||||
const char *host;
|
||||
const char *cachedir;
|
||||
const char *temp;
|
||||
const char *festivalcommand;
|
||||
int port=1314;
|
||||
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;
|
||||
char *newfestivalcommand;
|
||||
|
||||
if (ast_strlen_zero(vdata)) {
|
||||
ast_log(LOG_WARNING, "festival requires an argument (text)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
cfg = ast_config_load(FESTIVAL_CONFIG);
|
||||
if (!cfg) {
|
||||
ast_log(LOG_WARNING, "No such configuration file %s\n", FESTIVAL_CONFIG);
|
||||
ast_module_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";
|
||||
} else { /* This else parses the festivalcommand that we're sent from the config file for \n's, etc */
|
||||
int i, j;
|
||||
newfestivalcommand = alloca(strlen(festivalcommand) + 1);
|
||||
|
||||
for (i = 0, j = 0; i < strlen(festivalcommand); i++) {
|
||||
if (festivalcommand[i] == '\\' && festivalcommand[i + 1] == 'n') {
|
||||
newfestivalcommand[j++] = '\n';
|
||||
i++;
|
||||
} else if (festivalcommand[i] == '\\') {
|
||||
newfestivalcommand[j++] = festivalcommand[i + 1];
|
||||
i++;
|
||||
} else
|
||||
newfestivalcommand[j++] = festivalcommand[i];
|
||||
}
|
||||
newfestivalcommand[j] = '\0';
|
||||
festivalcommand = newfestivalcommand;
|
||||
}
|
||||
|
||||
data = ast_strdupa(vdata);
|
||||
|
||||
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);
|
||||
ast_module_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);
|
||||
ast_module_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);
|
||||
ast_module_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);
|
||||
ast_module_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);
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
struct ast_config *cfg = ast_config_load(FESTIVAL_CONFIG);
|
||||
if (!cfg) {
|
||||
ast_log(LOG_WARNING, "No such configuration file %s\n", FESTIVAL_CONFIG);
|
||||
return AST_MODULE_LOAD_DECLINE;
|
||||
}
|
||||
ast_config_destroy(cfg);
|
||||
return ast_register_application(app, festival_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Simple Festival Interface");
|
||||
124
apps/app_flash.c
124
apps/app_flash.c
@@ -1,124 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
/*** MODULEINFO
|
||||
<depend>zaptel</depend>
|
||||
***/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <errno.h>
|
||||
#include <sys/ioctl.h>
|
||||
#include <zaptel/zaptel.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/translate.h"
|
||||
#include "asterisk/image.h"
|
||||
#include "asterisk/options.h"
|
||||
|
||||
static char *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";
|
||||
|
||||
|
||||
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 ast_module_user *u;
|
||||
struct zt_params ztp;
|
||||
u = ast_module_user_add(chan);
|
||||
if (!strcasecmp(chan->tech->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);
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, flash_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Flash channel application");
|
||||
|
||||
1105
apps/app_followme.c
1105
apps/app_followme.c
File diff suppressed because it is too large
Load Diff
@@ -1,114 +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
|
||||
*
|
||||
* \author Anthony Minessale anthmct@yahoo.com
|
||||
*
|
||||
* \note Development of this app Sponsored/Funded by TAAN Softworks Corp
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/cdr.h"
|
||||
#include "asterisk/module.h"
|
||||
|
||||
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";
|
||||
|
||||
|
||||
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 ast_module_user *u;
|
||||
|
||||
if (!chan->cdr) {
|
||||
ast_log(LOG_WARNING, "Channel does not have a CDR\n");
|
||||
return 0;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (!ast_strlen_zero(data))
|
||||
ast_set2_flag(chan->cdr, strchr(data, 'v'), AST_CDR_FLAG_KEEP_VARS);
|
||||
|
||||
ast_cdr_fork(chan);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, forkcdr_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Fork The CDR into 2 separate entities");
|
||||
@@ -1,148 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/adsi.h"
|
||||
#include "asterisk/options.h"
|
||||
|
||||
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";
|
||||
|
||||
|
||||
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 ast_adsi_print(chan, tmp, justify, voice);
|
||||
}
|
||||
|
||||
static int cpeid_exec(struct ast_channel *chan, void *idata)
|
||||
{
|
||||
int res=0;
|
||||
struct ast_module_user *u;
|
||||
unsigned char cpeid[4];
|
||||
int gotgeometry = 0;
|
||||
int gotcpeid = 0;
|
||||
int width, height, buttons;
|
||||
char *data[4];
|
||||
unsigned int x;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
for (x = 0; x < 4; x++)
|
||||
data[x] = alloca(80);
|
||||
|
||||
strcpy(data[0], "** CPE Info **");
|
||||
strcpy(data[1], "Identifying CPE...");
|
||||
strcpy(data[2], "Please wait...");
|
||||
res = ast_adsi_load_session(chan, NULL, 0, 1);
|
||||
if (res > 0) {
|
||||
cpeid_setstatus(chan, data, 0);
|
||||
res = ast_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) {
|
||||
strcpy(data[1], "Measuring CPE...");
|
||||
strcpy(data[2], "Please wait...");
|
||||
cpeid_setstatus(chan, data, 0);
|
||||
res = ast_adsi_get_cpeinfo(chan, &width, &height, &buttons, 0);
|
||||
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(data[1], 80, "CPEID: %02x:%02x:%02x:%02x", cpeid[0], cpeid[1], cpeid[2], cpeid[3]);
|
||||
else
|
||||
strcpy(data[1], "CPEID Unknown");
|
||||
if (gotgeometry)
|
||||
snprintf(data[2], 80, "Geom: %dx%d, %d buttons", width, height, buttons);
|
||||
else
|
||||
strcpy(data[2], "Geometry unknown");
|
||||
strcpy(data[3], "Press # to exit");
|
||||
cpeid_setstatus(chan, data, 1);
|
||||
for(;;) {
|
||||
res = ast_waitfordigit(chan, 1000);
|
||||
if (res < 0)
|
||||
break;
|
||||
if (res == '#') {
|
||||
res = 0;
|
||||
break;
|
||||
}
|
||||
}
|
||||
ast_adsi_unload_session(chan);
|
||||
}
|
||||
}
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, cpeid_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Get ADSI CPE ID");
|
||||
@@ -1,222 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Changes Copyright (c) 2004 - 2006 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
|
||||
*
|
||||
* \author Todd Freeman <freeman@andrews.edu>
|
||||
*
|
||||
* \note 95% based on HasNewVoicemail by
|
||||
* Tilghman Lesher <asterisk-hasnewvoicemail-app@the-tilghman.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
#include <dirent.h>
|
||||
#include <sys/types.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/utils.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/options.h"
|
||||
|
||||
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"
|
||||
"\nThis application has been deprecated in favor of the VMCOUNT() function\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"
|
||||
"\nThis application has been deprecated in favor of the VMCOUNT() function\n";
|
||||
|
||||
|
||||
static int hasvoicemail_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *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;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
input = ast_strdupa(data);
|
||||
|
||||
AST_STANDARD_APP_ARGS(args, input);
|
||||
|
||||
vmbox = strsep(&args.vmbox, "@");
|
||||
|
||||
if (!ast_strlen_zero(args.vmbox))
|
||||
context = 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 = ast_app_messagecount(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 || ast_opt_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);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int acf_vmcount_exec(struct ast_channel *chan, char *cmd, char *argsstr, char *buf, size_t len)
|
||||
{
|
||||
struct ast_module_user *u;
|
||||
char *context;
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(vmbox);
|
||||
AST_APP_ARG(folder);
|
||||
);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
buf[0] = '\0';
|
||||
|
||||
AST_STANDARD_APP_ARGS(args, argsstr);
|
||||
|
||||
if (strchr(args.vmbox, '@')) {
|
||||
context = args.vmbox;
|
||||
args.vmbox = strsep(&context, "@");
|
||||
} else {
|
||||
context = "default";
|
||||
}
|
||||
|
||||
if (ast_strlen_zero(args.folder)) {
|
||||
args.folder = "INBOX";
|
||||
}
|
||||
|
||||
snprintf(buf, len, "%d", ast_app_messagecount(context, args.vmbox, args.folder));
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
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 unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_custom_function_unregister(&acf_vmcount);
|
||||
res |= ast_unregister_application(app_hasvoicemail);
|
||||
res |= ast_unregister_application(app_hasnewvoicemail);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
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;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Indicator for whether a voice mailbox has messages in a given folder.");
|
||||
223
apps/app_ices.c
223
apps/app_ices.c
@@ -1,223 +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)
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#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/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 *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";
|
||||
|
||||
|
||||
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 (ast_opt_high_priority)
|
||||
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 ast_module_user *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;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
last = ast_tv(0, 0);
|
||||
|
||||
if (pipe(fds)) {
|
||||
ast_log(LOG_WARNING, "Unable to create pipe\n");
|
||||
ast_module_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");
|
||||
ast_module_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");
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
if (((char *)data)[0] == '/')
|
||||
ast_copy_string(filename, (char *) data, sizeof(filename));
|
||||
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);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, ices_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Encode and Stream via icecast and ices");
|
||||
142
apps/app_image.c
142
apps/app_image.c
@@ -1,125 +1,89 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
* Asterisk -- A telephony toolkit for Linux.
|
||||
*
|
||||
* Copyright (C) 1999 - 2005, Digium, Inc.
|
||||
* App to transmit an image
|
||||
*
|
||||
* Copyright (C) 1999, Mark Spencer
|
||||
*
|
||||
* Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
* Mark Spencer <markster@linux-support.net>
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief App to transmit an image
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <asterisk/file.h>
|
||||
#include <asterisk/logger.h>
|
||||
#include <asterisk/channel.h>
|
||||
#include <asterisk/pbx.h>
|
||||
#include <asterisk/module.h>
|
||||
#include <asterisk/translate.h>
|
||||
#include <asterisk/image.h>
|
||||
#include <string.h>
|
||||
#include <stdlib.h>
|
||||
#include <pthread.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/translate.h"
|
||||
#include "asterisk/image.h"
|
||||
#include "asterisk/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";
|
||||
" SendImage(filename): Sends an image on a channel. If the channel\n"
|
||||
"does not support image transport, and there exists a step with\n"
|
||||
"priority n + 101, then execution will continue at that step.\n"
|
||||
"Otherwise, execution will continue at the next priority level.\n"
|
||||
"SendImage only returns 0 if the image was sent correctly or if\n"
|
||||
"the channel does not support image transport, and -1 otherwise.\n";
|
||||
|
||||
STANDARD_LOCAL_USER;
|
||||
|
||||
LOCAL_USER_DECL;
|
||||
|
||||
static int sendimage_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *u;
|
||||
char *parse;
|
||||
int priority_jump = 0;
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(filename);
|
||||
AST_APP_ARG(options);
|
||||
);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
parse = ast_strdupa(data);
|
||||
|
||||
AST_STANDARD_APP_ARGS(args, parse);
|
||||
|
||||
if (ast_strlen_zero(args.filename)) {
|
||||
ast_log(LOG_WARNING, "SendImage requires an argument (filename[|options])\n");
|
||||
struct localuser *u;
|
||||
if (!data || !strlen((char *)data)) {
|
||||
ast_log(LOG_WARNING, "SendImage requires an argument (filename)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
if (args.options) {
|
||||
if (strchr(args.options, 'j'))
|
||||
priority_jump = 1;
|
||||
}
|
||||
|
||||
LOCAL_USER_ADD(u);
|
||||
if (!ast_supports_images(chan)) {
|
||||
/* Does not support transport */
|
||||
if (priority_jump || ast_opt_priority_jumping)
|
||||
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
|
||||
pbx_builtin_setvar_helper(chan, "SENDIMAGESTATUS", "NOSUPPORT");
|
||||
ast_module_user_remove(u);
|
||||
if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
|
||||
chan->priority += 100;
|
||||
return 0;
|
||||
}
|
||||
|
||||
res = ast_send_image(chan, args.filename);
|
||||
|
||||
if (!res)
|
||||
pbx_builtin_setvar_helper(chan, "SENDIMAGESTATUS", "OK");
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
res = ast_send_image(chan, data);
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, sendimage_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Image Transmission Application");
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
|
||||
char *key()
|
||||
{
|
||||
return ASTERISK_GPL_KEY;
|
||||
}
|
||||
|
||||
196
apps/app_intercom.c
Normal file
196
apps/app_intercom.c
Normal file
@@ -0,0 +1,196 @@
|
||||
/*
|
||||
* Asterisk -- A telephony toolkit for Linux.
|
||||
*
|
||||
* Use /dev/dsp as an intercom.
|
||||
*
|
||||
* Copyright (C) 1999, Mark Spencer
|
||||
*
|
||||
* Mark Spencer <markster@linux-support.net>
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
#include <asterisk/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>
|
||||
|
||||
#define DEV_DSP "/dev/dsp"
|
||||
|
||||
/* Number of 32 byte buffers -- each buffer is 2 ms */
|
||||
#define BUFFER_SIZE 32
|
||||
|
||||
static char *tdesc = "Intercom using /dev/dsp for output";
|
||||
|
||||
static char *app = "Intercom";
|
||||
|
||||
static char *synopsis = "(Obsolete) Send to Intercom";
|
||||
static char *descrip =
|
||||
" Intercom(): Sends the user to the intercom (i.e. /dev/dsp). This program\n"
|
||||
"is generally considered obselete by the chan_oss module. Returns 0 if the\n"
|
||||
"user exits with a DTMF tone, or -1 if they hangup.\n";
|
||||
|
||||
STANDARD_LOCAL_USER;
|
||||
|
||||
LOCAL_USER_DECL;
|
||||
|
||||
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_pthread_mutex_lock(&sound_lock);
|
||||
if (sound < 0) {
|
||||
ast_log(LOG_WARNING, "Sound device closed?\n");
|
||||
ast_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_pthread_mutex_unlock(&sound_lock);
|
||||
return -1;
|
||||
}
|
||||
res = write(sound, data, len);
|
||||
ast_pthread_mutex_unlock(&sound_lock);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int create_audio()
|
||||
{
|
||||
int fmt, desired, res, fd;
|
||||
fd = open(DEV_DSP, O_WRONLY);
|
||||
if (fd < 0) {
|
||||
ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
|
||||
close(fd);
|
||||
return -1;
|
||||
}
|
||||
fmt = AFMT_S16_LE;
|
||||
res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
|
||||
if (res < 0) {
|
||||
ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
|
||||
close(fd);
|
||||
return -1;
|
||||
}
|
||||
fmt = 0;
|
||||
res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
|
||||
if (res < 0) {
|
||||
ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
|
||||
close(fd);
|
||||
return -1;
|
||||
}
|
||||
/* 8000 Hz desired */
|
||||
desired = 8000;
|
||||
fmt = desired;
|
||||
res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
|
||||
if (res < 0) {
|
||||
ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
|
||||
close(fd);
|
||||
return -1;
|
||||
}
|
||||
if (fmt != desired) {
|
||||
ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
|
||||
}
|
||||
#if 1
|
||||
/* 2 bytes * 15 units of 2^5 = 32 bytes per buffer */
|
||||
fmt = ((BUFFER_SIZE) << 16) | (0x0005);
|
||||
res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
|
||||
if (res < 0) {
|
||||
ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
|
||||
}
|
||||
#endif
|
||||
sound = fd;
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int intercom_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct localuser *u;
|
||||
struct ast_frame *f;
|
||||
int oreadformat;
|
||||
LOCAL_USER_ADD(u);
|
||||
/* Remember original read format */
|
||||
oreadformat = chan->readformat;
|
||||
/* Set mode to signed linear */
|
||||
res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
|
||||
if (res < 0) {
|
||||
ast_log(LOG_WARNING, "Unable to set format to signed linear on channel %s\n", chan->name);
|
||||
return -1;
|
||||
}
|
||||
/* Read packets from the channel */
|
||||
while(!res) {
|
||||
res = ast_waitfor(chan, -1);
|
||||
if (res > 0) {
|
||||
res = 0;
|
||||
f = ast_read(chan);
|
||||
if (f) {
|
||||
if (f->frametype == AST_FRAME_DTMF) {
|
||||
ast_frfree(f);
|
||||
break;
|
||||
} else {
|
||||
if (f->frametype == AST_FRAME_VOICE) {
|
||||
if (f->subclass == AST_FORMAT_SLINEAR) {
|
||||
res = write_audio(f->data, f->datalen);
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
} else
|
||||
ast_log(LOG_DEBUG, "Unable to handle non-signed linear frame (%d)\n", f->subclass);
|
||||
}
|
||||
}
|
||||
ast_frfree(f);
|
||||
} else
|
||||
res = -1;
|
||||
}
|
||||
}
|
||||
LOCAL_USER_REMOVE(u);
|
||||
if (!res)
|
||||
ast_set_read_format(chan, oreadformat);
|
||||
return res;
|
||||
}
|
||||
|
||||
int unload_module(void)
|
||||
{
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
if (sound > -1)
|
||||
close(sound);
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
int load_module(void)
|
||||
{
|
||||
if (create_audio())
|
||||
return -1;
|
||||
return ast_register_application(app, intercom_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
|
||||
char *key()
|
||||
{
|
||||
return ASTERISK_GPL_KEY;
|
||||
}
|
||||
@@ -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 IVR Demo application
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
/*** MODULEINFO
|
||||
<defaultenabled>no</defaultenabled>
|
||||
***/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/app.h"
|
||||
|
||||
static char *tdesc = "IVR Demo Application";
|
||||
static char *app = "IVRDemo";
|
||||
static char *synopsis =
|
||||
" This is a skeleton application that shows you the basic structure to create your\n"
|
||||
"own asterisk applications and demonstrates the IVR demo.\n";
|
||||
|
||||
static int ivr_demo_func(struct ast_channel *chan, void *data)
|
||||
{
|
||||
ast_verbose("IVR Demo, data is %s!\n", (char *)data);
|
||||
return 0;
|
||||
}
|
||||
|
||||
AST_IVR_DECLARE_MENU(ivr_submenu, "IVR Demo Sub Menu", 0,
|
||||
{
|
||||
{ "s", AST_ACTION_BACKGROUND, "demo-abouttotry" },
|
||||
{ "s", AST_ACTION_WAITOPTION },
|
||||
{ "1", AST_ACTION_PLAYBACK, "digits/1" },
|
||||
{ "1", AST_ACTION_PLAYBACK, "digits/1" },
|
||||
{ "1", AST_ACTION_RESTART },
|
||||
{ "2", AST_ACTION_PLAYLIST, "digits/2;digits/3" },
|
||||
{ "3", AST_ACTION_CALLBACK, ivr_demo_func },
|
||||
{ "4", AST_ACTION_TRANSFER, "demo|s|1" },
|
||||
{ "*", AST_ACTION_REPEAT },
|
||||
{ "#", AST_ACTION_UPONE },
|
||||
{ NULL }
|
||||
});
|
||||
|
||||
AST_IVR_DECLARE_MENU(ivr_demo, "IVR Demo Main Menu", 0,
|
||||
{
|
||||
{ "s", AST_ACTION_BACKGROUND, "demo-congrats" },
|
||||
{ "g", AST_ACTION_BACKGROUND, "demo-instruct" },
|
||||
{ "g", AST_ACTION_WAITOPTION },
|
||||
{ "1", AST_ACTION_PLAYBACK, "digits/1" },
|
||||
{ "1", AST_ACTION_RESTART },
|
||||
{ "2", AST_ACTION_MENU, &ivr_submenu },
|
||||
{ "2", AST_ACTION_RESTART },
|
||||
{ "i", AST_ACTION_PLAYBACK, "invalid" },
|
||||
{ "i", AST_ACTION_REPEAT, (void *)(unsigned long)2 },
|
||||
{ "#", AST_ACTION_EXIT },
|
||||
{ NULL },
|
||||
});
|
||||
|
||||
|
||||
static int skel_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res=0;
|
||||
struct ast_module_user *u;
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "skel requires an argument (filename)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
/* 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);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, skel_exec, tdesc, synopsis);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "IVR Demo Application");
|
||||
@@ -1,160 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
|
||||
#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 *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\n"
|
||||
"This application is deprecated and may be removed from a future release.\n"
|
||||
"Please use the dialplan function BLACKLIST() instead.\n";
|
||||
|
||||
|
||||
static int blacklist_read(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
|
||||
{
|
||||
char blacklist[1];
|
||||
int bl = 0;
|
||||
|
||||
if (chan->cid.cid_num) {
|
||||
if (!ast_db_get("blacklist", chan->cid.cid_num, blacklist, sizeof (blacklist)))
|
||||
bl = 1;
|
||||
}
|
||||
if (chan->cid.cid_name) {
|
||||
if (!ast_db_get("blacklist", chan->cid.cid_name, blacklist, sizeof (blacklist)))
|
||||
bl = 1;
|
||||
}
|
||||
|
||||
snprintf(buf, len, "%d", bl);
|
||||
return 0;
|
||||
}
|
||||
|
||||
static struct ast_custom_function blacklist_function = {
|
||||
.name = "BLACKLIST",
|
||||
.synopsis = "Check if the callerid is on the blacklist",
|
||||
.desc = "Uses astdb to check if the Caller*ID is in family 'blacklist'. Returns 1 or 0.\n",
|
||||
.syntax = "BLACKLIST()",
|
||||
.read = blacklist_read,
|
||||
};
|
||||
|
||||
static int
|
||||
lookupblacklist_exec (struct ast_channel *chan, void *data)
|
||||
{
|
||||
char blacklist[1];
|
||||
struct ast_module_user *u;
|
||||
int bl = 0;
|
||||
int priority_jump = 0;
|
||||
static int dep_warning = 0;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (!dep_warning) {
|
||||
dep_warning = 1;
|
||||
ast_log(LOG_WARNING, "LookupBlacklist is deprecated. Please use ${BLACKLIST()} instead.\n");
|
||||
}
|
||||
|
||||
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 || ast_opt_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");
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
res |= ast_custom_function_unregister(&blacklist_function);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
int res = ast_custom_function_register(&blacklist_function);
|
||||
res |= ast_register_application (app, lookupblacklist_exec, synopsis,descrip);
|
||||
return res;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Look up Caller*ID name/number from blacklist database");
|
||||
@@ -1,103 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
|
||||
#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 *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\n"
|
||||
"LookupCIDName is deprecated. Please use ${DB(cidname/${CALLERID(num)})}\n"
|
||||
"instead.\n";
|
||||
|
||||
|
||||
static int lookupcidname_exec (struct ast_channel *chan, void *data)
|
||||
{
|
||||
char dbname[64];
|
||||
struct ast_module_user *u;
|
||||
static int dep_warning = 0;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
if (!dep_warning) {
|
||||
dep_warning = 1;
|
||||
ast_log(LOG_WARNING, "LookupCIDName is deprecated. Please use ${DB(cidname/${CALLERID(num)})} instead.\n");
|
||||
}
|
||||
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);
|
||||
}
|
||||
}
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application (app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application (app, lookupcidname_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Look up CallerID Name from local database");
|
||||
557
apps/app_macro.c
557
apps/app_macro.c
@@ -1,557 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
#include <sys/types.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/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 *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 macros\n"
|
||||
" 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 *exclusive_descrip =
|
||||
" MacroExclusive(macroname|arg1|arg2...):\n"
|
||||
"Executes macro defined in the context 'macro-macroname'\n"
|
||||
"Only one call at a time may run the macro.\n"
|
||||
"(we'll wait if another call is busy executing in the Macro)\n"
|
||||
"Arguments and return values as in application Macro()\n";
|
||||
|
||||
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 *exclusive_app = "MacroExclusive";
|
||||
static char *exit_app = "MacroExit";
|
||||
|
||||
static char *synopsis = "Macro Implementation";
|
||||
static char *if_synopsis = "Conditional Macro Implementation";
|
||||
static char *exclusive_synopsis = "Exclusive Macro Implementation";
|
||||
static char *exit_synopsis = "Exit From Macro";
|
||||
|
||||
|
||||
static struct ast_exten *find_matching_priority(struct ast_context *c, const char *exten, int priority, const char *callerid)
|
||||
{
|
||||
struct ast_exten *e;
|
||||
struct ast_include *i;
|
||||
struct ast_context *c2;
|
||||
|
||||
for (e=ast_walk_context_extensions(c, NULL); e; e=ast_walk_context_extensions(c, e)) {
|
||||
if (ast_extension_match(ast_get_extension_name(e), exten)) {
|
||||
int needmatch = ast_get_extension_matchcid(e);
|
||||
if ((needmatch && ast_extension_match(ast_get_extension_cidmatch(e), callerid)) ||
|
||||
(!needmatch)) {
|
||||
/* This is the matching extension we want */
|
||||
struct ast_exten *p;
|
||||
for (p=ast_walk_extension_priorities(e, NULL); p; p=ast_walk_extension_priorities(e, p)) {
|
||||
if (priority != ast_get_extension_priority(p))
|
||||
continue;
|
||||
return p;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
/* No match; run through includes */
|
||||
for (i=ast_walk_context_includes(c, NULL); i; i=ast_walk_context_includes(c, i)) {
|
||||
for (c2=ast_walk_contexts(NULL); c2; c2=ast_walk_contexts(c2)) {
|
||||
if (!strcmp(ast_get_context_name(c2), ast_get_include_name(i))) {
|
||||
e = find_matching_priority(c2, exten, priority, callerid);
|
||||
if (e)
|
||||
return e;
|
||||
}
|
||||
}
|
||||
}
|
||||
return NULL;
|
||||
}
|
||||
|
||||
static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
|
||||
{
|
||||
const char *s;
|
||||
char *tmp;
|
||||
char *cur, *rest;
|
||||
char *macro;
|
||||
char fullmacro[80];
|
||||
char varname[80];
|
||||
char runningapp[80], runningdata[1024];
|
||||
char *oldargs[MAX_ARGS + 1] = { NULL, };
|
||||
int argc, x;
|
||||
int res=0;
|
||||
char oldexten[256]="";
|
||||
int oldpriority, gosub_level = 0;
|
||||
char pc[80], depthc[12];
|
||||
char oldcontext[AST_MAX_CONTEXT] = "";
|
||||
const char *inhangupc;
|
||||
int offset, depth = 0, maxdepth = 7;
|
||||
int setmacrocontext=0;
|
||||
int autoloopflag, dead = 0, inhangup = 0;
|
||||
|
||||
char *save_macro_exten;
|
||||
char *save_macro_context;
|
||||
char *save_macro_priority;
|
||||
char *save_macro_offset;
|
||||
struct ast_module_user *u;
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "Macro() requires arguments. See \"show application macro\" for help.\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
/* does the user want a deeper rabbit hole? */
|
||||
s = pbx_builtin_getvar_helper(chan, "MACRO_RECURSION");
|
||||
if (s)
|
||||
sscanf(s, "%d", &maxdepth);
|
||||
|
||||
/* Count how many levels deep the rabbit hole goes */
|
||||
s = pbx_builtin_getvar_helper(chan, "MACRO_DEPTH");
|
||||
if (s)
|
||||
sscanf(s, "%d", &depth);
|
||||
/* Used for detecting whether to return when a Macro is called from another Macro after hangup */
|
||||
if (strcmp(chan->exten, "h") == 0)
|
||||
pbx_builtin_setvar_helper(chan, "MACRO_IN_HANGUP", "1");
|
||||
inhangupc = pbx_builtin_getvar_helper(chan, "MACRO_IN_HANGUP");
|
||||
if (!ast_strlen_zero(inhangupc))
|
||||
sscanf(inhangupc, "%d", &inhangup);
|
||||
|
||||
if (depth >= maxdepth) {
|
||||
ast_log(LOG_ERROR, "Macro(): possible infinite loop detected. Returning early.\n");
|
||||
ast_module_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");
|
||||
ast_module_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);
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* If we are to run the macro exclusively, take the mutex */
|
||||
if (exclusive) {
|
||||
ast_log(LOG_DEBUG, "Locking macrolock for '%s'\n", fullmacro);
|
||||
ast_autoservice_start(chan);
|
||||
if (ast_context_lockmacro(fullmacro)) {
|
||||
ast_log(LOG_WARNING, "Failed to lock macro '%s' as in-use\n", fullmacro);
|
||||
ast_autoservice_stop(chan);
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
ast_autoservice_stop(chan);
|
||||
}
|
||||
|
||||
/* Save old info */
|
||||
oldpriority = chan->priority;
|
||||
ast_copy_string(oldexten, chan->exten, sizeof(oldexten));
|
||||
ast_copy_string(oldcontext, chan->context, sizeof(oldcontext));
|
||||
if (ast_strlen_zero(chan->macrocontext)) {
|
||||
ast_copy_string(chan->macrocontext, chan->context, sizeof(chan->macrocontext));
|
||||
ast_copy_string(chan->macroexten, chan->exten, sizeof(chan->macroexten));
|
||||
chan->macropriority = chan->priority;
|
||||
setmacrocontext=1;
|
||||
}
|
||||
argc = 1;
|
||||
/* Save old macro variables */
|
||||
save_macro_exten = ast_strdup(pbx_builtin_getvar_helper(chan, "MACRO_EXTEN"));
|
||||
pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", oldexten);
|
||||
|
||||
save_macro_context = ast_strdup(pbx_builtin_getvar_helper(chan, "MACRO_CONTEXT"));
|
||||
pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", oldcontext);
|
||||
|
||||
save_macro_priority = ast_strdup(pbx_builtin_getvar_helper(chan, "MACRO_PRIORITY"));
|
||||
snprintf(pc, sizeof(pc), "%d", oldpriority);
|
||||
pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", pc);
|
||||
|
||||
save_macro_offset = ast_strdup(pbx_builtin_getvar_helper(chan, "MACRO_OFFSET"));
|
||||
pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", NULL);
|
||||
|
||||
/* Setup environment for new run */
|
||||
chan->exten[0] = 's';
|
||||
chan->exten[1] = '\0';
|
||||
ast_copy_string(chan->context, fullmacro, sizeof(chan->context));
|
||||
chan->priority = 1;
|
||||
|
||||
while((cur = strsep(&rest, "|")) && (argc < MAX_ARGS)) {
|
||||
const char *s;
|
||||
/* Save copy of old arguments if we're overwriting some, otherwise
|
||||
let them pass through to the other macro */
|
||||
snprintf(varname, sizeof(varname), "ARG%d", argc);
|
||||
s = pbx_builtin_getvar_helper(chan, varname);
|
||||
if (s)
|
||||
oldargs[argc] = ast_strdup(s);
|
||||
pbx_builtin_setvar_helper(chan, varname, cur);
|
||||
argc++;
|
||||
}
|
||||
autoloopflag = ast_test_flag(chan, AST_FLAG_IN_AUTOLOOP);
|
||||
ast_set_flag(chan, AST_FLAG_IN_AUTOLOOP);
|
||||
while(ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num)) {
|
||||
struct ast_context *c;
|
||||
struct ast_exten *e;
|
||||
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);
|
||||
else if (option_verbose > 1)
|
||||
ast_verbose( VERBOSE_PREFIX_2 "Spawn extension (%s, %s, %d) exited KEEPALIVE in macro '%s' on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
|
||||
goto out;
|
||||
break;
|
||||
default:
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "Spawn extension (%s,%s,%d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
|
||||
else if (option_verbose > 1)
|
||||
ast_verbose( VERBOSE_PREFIX_2 "Spawn extension (%s, %s, %d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
|
||||
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 */
|
||||
const char *offsets;
|
||||
ast_copy_string(chan->exten, oldexten, sizeof(chan->exten));
|
||||
if ((offsets = pbx_builtin_getvar_helper(chan, "MACRO_OFFSET"))) {
|
||||
/* Handle macro offset if it's set by checking the availability of step n + offset + 1, otherwise continue
|
||||
normally if there is any problem */
|
||||
if (sscanf(offsets, "%d", &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);
|
||||
|
||||
/* Unlock the macro */
|
||||
if (exclusive) {
|
||||
ast_log(LOG_DEBUG, "Unlocking macrolock for '%s'\n", fullmacro);
|
||||
if (ast_context_unlockmacro(fullmacro)) {
|
||||
ast_log(LOG_ERROR, "Failed to unlock macro '%s' - that isn't good\n", fullmacro);
|
||||
res = 0;
|
||||
}
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int macro_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
return _macro_exec(chan, data, 0);
|
||||
}
|
||||
|
||||
static int macroexclusive_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
return _macro_exec(chan, data, 1);
|
||||
}
|
||||
|
||||
static int macroif_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
char *expr = NULL, *label_a = NULL, *label_b = NULL;
|
||||
int res = 0;
|
||||
struct ast_module_user *u;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (!(expr = ast_strdupa(data))) {
|
||||
ast_module_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))
|
||||
macro_exec(chan, label_a);
|
||||
else if (label_b)
|
||||
macro_exec(chan, label_b);
|
||||
} else
|
||||
ast_log(LOG_WARNING, "Invalid Syntax.\n");
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int macro_exit_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
return MACRO_EXIT_RESULT;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(if_app);
|
||||
res |= ast_unregister_application(exit_app);
|
||||
res |= ast_unregister_application(app);
|
||||
res |= ast_unregister_application(exclusive_app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_register_application(exit_app, macro_exit_exec, exit_synopsis, exit_descrip);
|
||||
res |= ast_register_application(if_app, macroif_exec, if_synopsis, if_descrip);
|
||||
res |= ast_register_application(exclusive_app, macroexclusive_exec, exclusive_synopsis, exclusive_descrip);
|
||||
res |= ast_register_application(app, macro_exec, synopsis, descrip);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Extension Macros");
|
||||
4820
apps/app_meetme.c
4820
apps/app_meetme.c
File diff suppressed because it is too large
Load Diff
@@ -1,153 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
#include <errno.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/utils.h"
|
||||
|
||||
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";
|
||||
|
||||
|
||||
static char digital_milliwatt[] = {0x1e,0x0b,0x0b,0x1e,0x9e,0x8b,0x8b,0x9e} ;
|
||||
|
||||
static void *milliwatt_alloc(struct ast_channel *chan, void *params)
|
||||
{
|
||||
return ast_calloc(1, sizeof(int));
|
||||
}
|
||||
|
||||
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)
|
||||
{
|
||||
unsigned char buf[AST_FRIENDLY_OFFSET + 640];
|
||||
const int maxsamples = sizeof (buf) / sizeof (buf[0]);
|
||||
int i, *indexp = (int *) data;
|
||||
struct ast_frame wf = {
|
||||
.frametype = AST_FRAME_VOICE,
|
||||
.subclass = AST_FORMAT_ULAW,
|
||||
.offset = AST_FRIENDLY_OFFSET,
|
||||
.data = buf + AST_FRIENDLY_OFFSET,
|
||||
.src = __FUNCTION__,
|
||||
};
|
||||
|
||||
/* Instead of len, use samples, because channel.c generator_force
|
||||
* generate(chan, tmp, 0, 160) ignores len. In any case, len is
|
||||
* a multiple of samples, given by number of samples times bytes per
|
||||
* sample. In the case of ulaw, len = samples. for signed linear
|
||||
* len = 2 * samples */
|
||||
|
||||
if (samples > maxsamples) {
|
||||
ast_log(LOG_WARNING, "Only doing %d samples (%d requested)\n", maxsamples, samples);
|
||||
samples = maxsamples;
|
||||
}
|
||||
len = samples * sizeof (buf[0]);
|
||||
wf.datalen = len;
|
||||
wf.samples = samples;
|
||||
/* 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 ast_module_user *u;
|
||||
u = ast_module_user_add(chan);
|
||||
ast_set_write_format(chan, AST_FORMAT_ULAW);
|
||||
ast_set_read_format(chan, AST_FORMAT_ULAW);
|
||||
if (chan->_state != AST_STATE_UP)
|
||||
{
|
||||
ast_answer(chan);
|
||||
}
|
||||
if (ast_activate_generator(chan,&milliwattgen,"milliwatt") < 0)
|
||||
{
|
||||
ast_log(LOG_WARNING,"Failed to activate generator on '%s'\n",chan->name);
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
while(!ast_safe_sleep(chan, 10000));
|
||||
ast_deactivate_generator(chan);
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, milliwatt_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Digital Milliwatt (mu-law) Test Application");
|
||||
@@ -1,463 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 2005, Anthony Minessale II
|
||||
* Copyright (C) 2005 - 2006, Digium, Inc.
|
||||
*
|
||||
* Mark Spencer <markster@digium.com>
|
||||
* Kevin P. Fleming <kpfleming@digium.com>
|
||||
*
|
||||
* Based on app_muxmon.c provided by
|
||||
* Anthony Minessale II <anthmct@yahoo.com>
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief MixMonitor() - Record a call and mix the audio during the recording
|
||||
* \ingroup applications
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
* \author Kevin P. Fleming <kpfleming@digium.com>
|
||||
*
|
||||
* \note Based on app_muxmon.c provided by
|
||||
* Anthony Minessale II <anthmct@yahoo.com>
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/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"
|
||||
#include "asterisk/utils.h"
|
||||
|
||||
#define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0
|
||||
|
||||
static const char *app = "MixMonitor";
|
||||
static const char *synopsis = "Record a call and mix the audio during the recording";
|
||||
static const char *desc = ""
|
||||
" MixMonitor(<file>.<ext>[|<options>[|<command>]])\n\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} and \n"
|
||||
"all variables will be evaluated at that time.\n"
|
||||
"The variable MIXMONITOR_FILENAME will contain the filename used to record.\n"
|
||||
"";
|
||||
|
||||
static const char *stop_app = "StopMixMonitor";
|
||||
static const char *stop_synopsis = "Stop recording a call through MixMonitor";
|
||||
static const char *stop_desc = ""
|
||||
" StopMixMonitor()\n\n"
|
||||
"Stops the audio recording that was started with a call to MixMonitor()\n"
|
||||
"on the current channel.\n"
|
||||
"";
|
||||
|
||||
struct module_symbols *me;
|
||||
|
||||
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_channel_lock(chan);
|
||||
res = ast_channel_spy_add(chan, spy);
|
||||
ast_channel_unlock(chan);
|
||||
|
||||
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;
|
||||
|
||||
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 = AST_LIST_NEXT(f, frame_list);
|
||||
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_frame_free(f, 0);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
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);
|
||||
|
||||
|
||||
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))) {
|
||||
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_background(&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 ast_module_user *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;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
parse = ast_strdupa(data);
|
||||
|
||||
AST_STANDARD_APP_ARGS(args, parse);
|
||||
|
||||
if (ast_strlen_zero(args.filename)) {
|
||||
ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
|
||||
ast_module_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], "%d", &x) != 1) || (x < -4) || (x > 4)) {
|
||||
ast_log(LOG_NOTICE, "Heard volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_READVOLUME]);
|
||||
} else {
|
||||
readvol = get_volfactor(x);
|
||||
}
|
||||
}
|
||||
|
||||
if (ast_test_flag(&flags, MUXFLAG_WRITEVOLUME)) {
|
||||
if (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], "%d", &x) != 1) || (x < -4) || (x > 4)) {
|
||||
ast_log(LOG_NOTICE, "Spoken volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_WRITEVOLUME]);
|
||||
} else {
|
||||
writevol = get_volfactor(x);
|
||||
}
|
||||
}
|
||||
|
||||
if (ast_test_flag(&flags, MUXFLAG_VOLUME)) {
|
||||
if (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], "%d", &x) != 1) || (x < -4) || (x > 4)) {
|
||||
ast_log(LOG_NOTICE, "Combined volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_VOLUME]);
|
||||
} else {
|
||||
readvol = writevol = get_volfactor(x);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
/* 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);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int stop_mixmonitor_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *u;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
ast_channel_lock(chan);
|
||||
ast_channel_spy_stop_by_type(chan, mixmonitor_spy_type);
|
||||
ast_channel_unlock(chan);
|
||||
|
||||
ast_module_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_channel_unlock(chan);
|
||||
|
||||
return RESULT_SUCCESS;
|
||||
}
|
||||
|
||||
static char *complete_mixmonitor_cli(const char *line, const char *word, int pos, int state)
|
||||
{
|
||||
return ast_complete_channels(line, word, pos, state, 2);
|
||||
}
|
||||
|
||||
static struct ast_cli_entry cli_mixmonitor[] = {
|
||||
{ { "mixmonitor", NULL, NULL },
|
||||
mixmonitor_cli, "Execute a MixMonitor command.",
|
||||
"mixmonitor <start|stop> <chan_name> [args]\n\n"
|
||||
"The optional arguments are passed to the\n"
|
||||
"MixMonitor application when the 'start' command is used.\n",
|
||||
complete_mixmonitor_cli },
|
||||
};
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
ast_cli_unregister_multiple(cli_mixmonitor, sizeof(cli_mixmonitor) / sizeof(struct ast_cli_entry));
|
||||
res = ast_unregister_application(stop_app);
|
||||
res |= ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
ast_cli_register_multiple(cli_mixmonitor, sizeof(cli_mixmonitor) / sizeof(struct ast_cli_entry));
|
||||
res = ast_register_application(app, mixmonitor_exec, synopsis, desc);
|
||||
res |= ast_register_application(stop_app, stop_mixmonitor_exec, stop_synopsis, stop_desc);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Mixed Audio Monitoring Application");
|
||||
@@ -1,178 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (c) 2006, Tilghman Lesher. All rights reserved.
|
||||
*
|
||||
* Tilghman Lesher <app_morsecode__v001@the-tilghman.com>
|
||||
*
|
||||
* This code is released by the author with no restrictions on usage.
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief Morsecode application
|
||||
*
|
||||
* \author Tilghman Lesher <app_morsecode__v001@the-tilghman.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/indications.h"
|
||||
|
||||
static char *app_morsecode = "Morsecode";
|
||||
|
||||
static char *morsecode_synopsis = "Plays morse code";
|
||||
|
||||
static char *morsecode_descrip =
|
||||
"Usage: Morsecode(<string>)\n"
|
||||
"Plays the Morse code equivalent of the passed string. If the variable\n"
|
||||
"MORSEDITLEN is set, it will use that value for the length (in ms) of the dit\n"
|
||||
"(defaults to 80). Additionally, if MORSETONE is set, it will use that tone\n"
|
||||
"(in Hz). The tone default is 800.\n";
|
||||
|
||||
|
||||
static char *morsecode[] = {
|
||||
"", "", "", "", "", "", "", "", "", "", "", "", "", "", "", "", /* 0-15 */
|
||||
"", "", "", "", "", "", "", "", "", "", "", "", "", "", "", "", /* 16-31 */
|
||||
" ", /* 32 - <space> */
|
||||
".-.-.-", /* 33 - ! */
|
||||
".-..-.", /* 34 - " */
|
||||
"", /* 35 - # */
|
||||
"", /* 36 - $ */
|
||||
"", /* 37 - % */
|
||||
"", /* 38 - & */
|
||||
".----.", /* 39 - ' */
|
||||
"-.--.-", /* 40 - ( */
|
||||
"-.--.-", /* 41 - ) */
|
||||
"", /* 42 - * */
|
||||
"", /* 43 - + */
|
||||
"--..--", /* 44 - , */
|
||||
"-....-", /* 45 - - */
|
||||
".-.-.-", /* 46 - . */
|
||||
"-..-.", /* 47 - / */
|
||||
"-----", ".----", "..---", "...--", "....-", ".....", "-....", "--...", "---..", "----.", /* 48-57 - 0-9 */
|
||||
"---...", /* 58 - : */
|
||||
"-.-.-.", /* 59 - ; */
|
||||
"", /* 60 - < */
|
||||
"-...-", /* 61 - = */
|
||||
"", /* 62 - > */
|
||||
"..--..", /* 63 - ? */
|
||||
".--.-.", /* 64 - @ */
|
||||
".-", "-...", "-.-.", "-..", ".", "..-.", "--.", "....", "..", ".---", "-.-", ".-..", "--",
|
||||
"-.", "---", ".--.", "--.-", ".-.", "...", "-", "..-", "...-", ".--", "-..-", "-.--", "--..",
|
||||
"-.--.-", /* 91 - [ (really '(') */
|
||||
"-..-.", /* 92 - \ (really '/') */
|
||||
"-.--.-", /* 93 - ] (really ')') */
|
||||
"", /* 94 - ^ */
|
||||
"..--.-", /* 95 - _ */
|
||||
".----.", /* 96 - ` */
|
||||
".-", "-...", "-.-.", "-..", ".", "..-.", "--.", "....", "..", ".---", "-.-", ".-..", "--",
|
||||
"-.", "---", ".--.", "--.-", ".-.", "...", "-", "..-", "...-", ".--", "-..-", "-.--", "--..",
|
||||
"-.--.-", /* 123 - { (really '(') */
|
||||
"", /* 124 - | */
|
||||
"-.--.-", /* 125 - } (really ')') */
|
||||
"-..-.", /* 126 - ~ (really bar) */
|
||||
". . .", /* 127 - <del> (error) */
|
||||
};
|
||||
|
||||
static void playtone(struct ast_channel *chan, int tone, int len)
|
||||
{
|
||||
char dtmf[20];
|
||||
snprintf(dtmf, sizeof(dtmf), "%d/%d", tone, len);
|
||||
ast_playtones_start(chan, 0, dtmf, 0);
|
||||
ast_safe_sleep(chan, len);
|
||||
ast_playtones_stop(chan);
|
||||
}
|
||||
|
||||
static int morsecode_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res=0, ditlen, tone;
|
||||
char *digit;
|
||||
const char *ditlenc, *tonec;
|
||||
struct ast_module_user *u;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "Syntax: Morsecode(<string>) - no argument found\n");
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* Use variable MORESEDITLEN, if set (else 80) */
|
||||
ditlenc = pbx_builtin_getvar_helper(chan, "MORSEDITLEN");
|
||||
if (ast_strlen_zero(ditlenc) || (sscanf(ditlenc, "%d", &ditlen) != 1)) {
|
||||
ditlen = 80;
|
||||
}
|
||||
|
||||
/* Use variable MORSETONE, if set (else 800) */
|
||||
tonec = pbx_builtin_getvar_helper(chan, "MORSETONE");
|
||||
if (ast_strlen_zero(tonec) || (sscanf(tonec, "%d", &tone) != 1)) {
|
||||
tone = 800;
|
||||
}
|
||||
|
||||
for (digit = data; *digit; digit++) {
|
||||
char *dahdit;
|
||||
if (*digit < 0) {
|
||||
continue;
|
||||
}
|
||||
for (dahdit = morsecode[(int)*digit]; *dahdit; dahdit++) {
|
||||
if (*dahdit == '-') {
|
||||
playtone(chan, tone, 3 * ditlen);
|
||||
} else if (*dahdit == '.') {
|
||||
playtone(chan, tone, 1 * ditlen);
|
||||
} else {
|
||||
/* Account for ditlen of silence immediately following */
|
||||
playtone(chan, 0, 2 * ditlen);
|
||||
}
|
||||
|
||||
/* Pause slightly between each dit and dah */
|
||||
playtone(chan, 0, 1 * ditlen);
|
||||
}
|
||||
/* Pause between characters */
|
||||
playtone(chan, 0, 2 * ditlen);
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app_morsecode);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app_morsecode, morsecode_exec, morsecode_synopsis, morsecode_descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Morse code");
|
||||
257
apps/app_mp3.c
257
apps/app_mp3.c
@@ -1,192 +1,172 @@
|
||||
/*
|
||||
* 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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#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/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.";
|
||||
" MP3Player(location) Executes mpg123 to play the given location\n"
|
||||
"which typically would be a filename or a URL. Returns -1 on\n"
|
||||
"hangup or 0 otherwise. User can exit by pressing any key\n.";
|
||||
|
||||
STANDARD_LOCAL_USER;
|
||||
|
||||
LOCAL_USER_DECL;
|
||||
|
||||
static int mp3play(char *filename, int fd)
|
||||
{
|
||||
int res;
|
||||
int x;
|
||||
sigset_t fullset, oldset;
|
||||
|
||||
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 (ast_opt_high_priority)
|
||||
ast_set_priority(0);
|
||||
signal(SIGPIPE, SIG_DFL);
|
||||
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
|
||||
|
||||
dup2(fd, STDOUT_FILENO);
|
||||
for (x=STDERR_FILENO + 1;x<256;x++) {
|
||||
for (x=0;x<256;x++) {
|
||||
if (x != STDOUT_FILENO)
|
||||
close(x);
|
||||
}
|
||||
/* Execute mpg123, but buffer if it's a net connection */
|
||||
if (!strncasecmp(filename, "http://", 7)) {
|
||||
/* 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 ast_module_user *u;
|
||||
struct localuser *u;
|
||||
int fds[2];
|
||||
int rfds[1 + AST_MAX_FDS];
|
||||
int ms = -1;
|
||||
int pid = -1;
|
||||
int us;
|
||||
int exception;
|
||||
int owriteformat;
|
||||
int timeout = 2000;
|
||||
struct timeval next;
|
||||
struct timeval tv;
|
||||
struct timeval last;
|
||||
struct ast_frame *f;
|
||||
int x;
|
||||
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;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (pipe(fds)) {
|
||||
ast_log(LOG_WARNING, "Unable to create pipe\n");
|
||||
ast_module_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");
|
||||
ast_module_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 */
|
||||
rfds[AST_MAX_FDS] = fds[0];
|
||||
for (;;) {
|
||||
ms = ast_tvdiff_ms(next, ast_tvnow());
|
||||
if (ms <= 0) {
|
||||
res = timed_read(fds[0], myf.frdata, sizeof(myf.frdata), timeout);
|
||||
CHECK_BLOCKING(chan);
|
||||
for (x=0;x<AST_MAX_FDS;x++)
|
||||
rfds[x] = chan->fds[x];
|
||||
res = ast_waitfor_n_fd(rfds, AST_MAX_FDS+1, &ms, &exception);
|
||||
chan->blocking = 0;
|
||||
if (res < 1) {
|
||||
ast_log(LOG_DEBUG, "Hangup detected\n");
|
||||
res = -1;
|
||||
break;
|
||||
}
|
||||
for(x=0;x<AST_MAX_FDS;x++)
|
||||
if (res == chan->fds[x])
|
||||
break;
|
||||
|
||||
if (x < AST_MAX_FDS) {
|
||||
if (exception)
|
||||
chan->exception = 1;
|
||||
f = ast_read(chan);
|
||||
if (!f) {
|
||||
ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
|
||||
res = -1;
|
||||
break;
|
||||
}
|
||||
if (f->frametype == AST_FRAME_DTMF) {
|
||||
ast_log(LOG_DEBUG, "User pressed a key\n");
|
||||
ast_frfree(f);
|
||||
res = 0;
|
||||
break;
|
||||
}
|
||||
ast_frfree(f);
|
||||
} else if (res == fds[0]) {
|
||||
gettimeofday(&tv, NULL);
|
||||
if (last.tv_sec || last.tv_usec) {
|
||||
/* We should wait at least a frame length */
|
||||
us = sizeof(myf.frdata) / 16 * 1000;
|
||||
/* Subtract 1,000,000 us for each second late we've passed */
|
||||
us -= (tv.tv_sec - last.tv_sec) * 1000000;
|
||||
/* And one for each us late we've passed */
|
||||
us -= (tv.tv_usec - last.tv_usec);
|
||||
/* Sleep that long if needed */
|
||||
if (us > 0)
|
||||
usleep(us);
|
||||
}
|
||||
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.samples = res / 2;
|
||||
myf.f.timelen = res / 16;
|
||||
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;
|
||||
@@ -195,61 +175,48 @@ static int mp3_exec(struct ast_channel *chan, void *data)
|
||||
} 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);
|
||||
}
|
||||
ast_log(LOG_DEBUG, "HuhHHH?\n");
|
||||
res = -1;
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
close(fds[0]);
|
||||
close(fds[1]);
|
||||
|
||||
LOCAL_USER_REMOVE(u);
|
||||
if (pid > -1)
|
||||
kill(pid, SIGKILL);
|
||||
if (!res && owriteformat)
|
||||
ast_set_write_format(chan, owriteformat);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, mp3_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Silly MP3 Application");
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
|
||||
char *key()
|
||||
{
|
||||
return ASTERISK_GPL_KEY;
|
||||
}
|
||||
|
||||
@@ -1,237 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#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/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 *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.";
|
||||
|
||||
|
||||
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 (ast_opt_high_priority)
|
||||
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 ast_module_user *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;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (socketpair(AF_LOCAL, SOCK_STREAM, 0, fds)) {
|
||||
ast_log(LOG_WARNING, "Unable to create socketpair\n");
|
||||
ast_module_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");
|
||||
ast_module_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);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, NBScat_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Silly NBS Stream Application");
|
||||
1649
apps/app_osplookup.c
1649
apps/app_osplookup.c
File diff suppressed because it is too large
Load Diff
203
apps/app_page.c
203
apps/app_page.c
@@ -1,203 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (c) 2004 - 2006 Digium, Inc. All rights reserved.
|
||||
*
|
||||
* Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* This code is released under the GNU General Public License
|
||||
* version 2.0. See LICENSE for more information.
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief page() - Paging application
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
/*** MODULEINFO
|
||||
<depend>zaptel</depend>
|
||||
<depend>app_meetme</depend>
|
||||
***/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
#include <errno.h>
|
||||
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/chanvars.h"
|
||||
#include "asterisk/utils.h"
|
||||
#include "asterisk/dial.h"
|
||||
#include "asterisk/devicestate.h"
|
||||
|
||||
static const char *app_page= "Page";
|
||||
|
||||
static const char *page_synopsis = "Pages phones";
|
||||
|
||||
static const char *page_descrip =
|
||||
"Page(Technology/Resource&Technology2/Resource2[|options])\n"
|
||||
" Places outbound calls to the given technology / resource and dumps\n"
|
||||
"them into a conference bridge as muted participants. The original\n"
|
||||
"caller is dumped into the conference as a speaker and the room is\n"
|
||||
"destroyed when the original caller leaves. Valid options are:\n"
|
||||
" d - full duplex audio\n"
|
||||
" q - quiet, do not play beep to caller\n"
|
||||
" r - record the page into a file (see 'r' for app_meetme)\n";
|
||||
|
||||
enum {
|
||||
PAGE_DUPLEX = (1 << 0),
|
||||
PAGE_QUIET = (1 << 1),
|
||||
PAGE_RECORD = (1 << 2),
|
||||
} page_opt_flags;
|
||||
|
||||
AST_APP_OPTIONS(page_opts, {
|
||||
AST_APP_OPTION('d', PAGE_DUPLEX),
|
||||
AST_APP_OPTION('q', PAGE_QUIET),
|
||||
AST_APP_OPTION('r', PAGE_RECORD),
|
||||
});
|
||||
|
||||
#define MAX_DIALS 128
|
||||
|
||||
static int page_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *u;
|
||||
char *options, *tech, *resource, *tmp;
|
||||
char meetmeopts[88], originator[AST_CHANNEL_NAME];
|
||||
struct ast_flags flags = { 0 };
|
||||
unsigned int confid = ast_random();
|
||||
struct ast_app *app;
|
||||
int res = 0, pos = 0, i = 0;
|
||||
struct ast_dial *dials[MAX_DIALS];
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (!(app = pbx_findapp("MeetMe"))) {
|
||||
ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
};
|
||||
|
||||
options = ast_strdupa(data);
|
||||
|
||||
ast_copy_string(originator, chan->name, sizeof(originator));
|
||||
if ((tmp = strchr(originator, '-')))
|
||||
*tmp = '\0';
|
||||
|
||||
tmp = strsep(&options, "|");
|
||||
if (options)
|
||||
ast_app_parse_options(page_opts, &flags, NULL, options);
|
||||
|
||||
snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe|%ud|%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
|
||||
(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
|
||||
|
||||
/* Go through parsing/calling each device */
|
||||
while ((tech = strsep(&tmp, "&"))) {
|
||||
struct ast_dial *dial = NULL;
|
||||
|
||||
/* don't call the originating device */
|
||||
if (!strcasecmp(tech, originator))
|
||||
continue;
|
||||
|
||||
/* If no resource is available, continue on */
|
||||
if (!(resource = strchr(tech, '/'))) {
|
||||
ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
|
||||
continue;
|
||||
}
|
||||
|
||||
*resource++ = '\0';
|
||||
|
||||
/* Create a dialing structure */
|
||||
if (!(dial = ast_dial_create())) {
|
||||
ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
|
||||
continue;
|
||||
}
|
||||
|
||||
/* Append technology and resource */
|
||||
ast_dial_append(dial, tech, resource);
|
||||
|
||||
/* Set ANSWER_EXEC as global option */
|
||||
ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
|
||||
|
||||
/* Run this dial in async mode */
|
||||
ast_dial_run(dial, chan, 1);
|
||||
|
||||
/* Put in our dialing array */
|
||||
dials[pos++] = dial;
|
||||
}
|
||||
|
||||
if (!ast_test_flag(&flags, PAGE_QUIET)) {
|
||||
res = ast_streamfile(chan, "beep", chan->language);
|
||||
if (!res)
|
||||
res = ast_waitstream(chan, "");
|
||||
}
|
||||
|
||||
if (!res) {
|
||||
snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"),
|
||||
(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
|
||||
pbx_exec(chan, app, meetmeopts);
|
||||
}
|
||||
|
||||
/* Go through each dial attempt cancelling, joining, and destroying */
|
||||
for (i = 0; i < pos; i++) {
|
||||
struct ast_dial *dial = dials[i];
|
||||
|
||||
/* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
|
||||
ast_dial_join(dial);
|
||||
|
||||
/* Hangup all channels */
|
||||
ast_dial_hangup(dial);
|
||||
|
||||
/* Destroy dialing structure */
|
||||
ast_dial_destroy(dial);
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return -1;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app_page);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");
|
||||
|
||||
@@ -1,260 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 1999 - 2006, 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
|
||||
*
|
||||
* \author Ben Miller <bgmiller@dccinc.com>
|
||||
* \arg With TONS of help from Mark!
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
#include <sys/types.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/features.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/say.h"
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/utils.h"
|
||||
|
||||
static char *app = "ParkAndAnnounce";
|
||||
|
||||
static char *synopsis = "Park and Announce";
|
||||
|
||||
static char *descrip =
|
||||
" ParkAndAnnounce(announce:template|timeout|dial|[return_context]):\n"
|
||||
"Park a call into the parkinglot and announce the call to another channel.\n"
|
||||
"\n"
|
||||
"announce template: Colon-separated list of files to announce. The word PARKED\n"
|
||||
" will be replaced by a say_digits of the extension in which\n"
|
||||
" the call is parked.\n"
|
||||
"timeout: Time in seconds before the call returns into the return\n"
|
||||
" context.\n"
|
||||
"dial: The app_dial style resource to call to make the\n"
|
||||
" announcement. Console/dsp calls the console.\n"
|
||||
"return_context: The goto-style label to jump the call back into after\n"
|
||||
" timeout. Default <priority+1>.\n"
|
||||
"\n"
|
||||
"The variable ${PARKEDAT} will contain the parking extension into which the\n"
|
||||
"call was placed. Use with the Local channel to allow the dialplan to make\n"
|
||||
"use of this information.\n";
|
||||
|
||||
|
||||
static int parkandannounce_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res=0;
|
||||
char *return_context;
|
||||
int lot, timeout = 0, dres;
|
||||
char *working, *context, *exten, *priority, *dial, *dialtech, *dialstr;
|
||||
char *template, *tpl_working, *tpl_current;
|
||||
char *tmp[100];
|
||||
char buf[13];
|
||||
int looptemp=0,i=0;
|
||||
char *s;
|
||||
|
||||
struct ast_channel *dchan;
|
||||
struct outgoing_helper oh;
|
||||
int outstate;
|
||||
|
||||
struct ast_module_user *u;
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "ParkAndAnnounce requires arguments: (announce:template|timeout|dial|[return_context])\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
s = ast_strdupa(data);
|
||||
|
||||
template=strsep(&s,"|");
|
||||
if(! template) {
|
||||
ast_log(LOG_WARNING, "PARK: An announce template must be defined\n");
|
||||
ast_module_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");
|
||||
ast_module_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);
|
||||
ast_module_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)
|
||||
ast_copy_string(chan->exten, exten, sizeof(chan->exten));
|
||||
if (context)
|
||||
ast_copy_string(chan->context, context, sizeof(chan->context));
|
||||
} 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 */
|
||||
|
||||
snprintf(buf, sizeof(buf), "%d", lot);
|
||||
memset(&oh, 0, sizeof(oh));
|
||||
oh.parent_channel = chan;
|
||||
oh.vars = ast_variable_new("_PARKEDAT", buf);
|
||||
dchan = __ast_request_and_dial(dialtech, AST_FORMAT_SLINEAR, 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);
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
} else {
|
||||
ast_log(LOG_WARNING, "PARK: Unable to allocate announce channel.\n");
|
||||
ast_module_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);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
/* return ast_register_application(app, park_exec); */
|
||||
return ast_register_application(app, parkandannounce_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Call Parking and Announce Application");
|
||||
@@ -1,492 +1,113 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
* Asterisk -- A telephony toolkit for Linux.
|
||||
*
|
||||
* Copyright (C) 1999 - 2005, Digium, Inc.
|
||||
* Trivial application to playback a sound file
|
||||
*
|
||||
* Copyright (C) 1999, Mark Spencer
|
||||
*
|
||||
* Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
* Mark Spencer <markster@linux-support.net>
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief Trivial application to playback a sound file
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
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/translate.h>
|
||||
#include <string.h>
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <pthread.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/translate.h"
|
||||
#include "asterisk/utils.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/cli.h"
|
||||
#include "asterisk/localtime.h"
|
||||
#include "asterisk/say.h"
|
||||
static char *tdesc = "Trivial Playback Application";
|
||||
|
||||
static char *app = "Playback";
|
||||
|
||||
static char *synopsis = "Play a file";
|
||||
|
||||
static char *descrip =
|
||||
" Playback(filename[&filename2...][|option]): Plays back given filenames (do not put\n"
|
||||
"extension). Options may also be included following a 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"
|
||||
;
|
||||
" Playback(filename[|option]): Plays back a given filename (do not put\n"
|
||||
"extension). Options may also be included following a pipe symbol. The only\n"
|
||||
"defined option at this time is 'skip', which causes the playback of the\n"
|
||||
"message to be skipped if the channel is not in the 'up' state (i.e. it\n"
|
||||
"hasn't been answered yet. If 'skip' is specified, the application will\n"
|
||||
"return immediately should the channel not be off hook. Otherwise, unless\n"
|
||||
"'noanswer' is specified, the channel channel will be answered before the sound\n"
|
||||
"is played. Not all channels support playing messages while on hook. Returns -1\n"
|
||||
"if the channel was hung up, or if the file does not exist. Returns 0 otherwise.\n";
|
||||
|
||||
STANDARD_LOCAL_USER;
|
||||
|
||||
static struct ast_config *say_cfg = NULL;
|
||||
/* save the say' api calls.
|
||||
* The first entry is NULL if we have the standard source,
|
||||
* otherwise we are sourcing from here.
|
||||
* 'say load [new|old]' will enable the new or old method, or report status
|
||||
*/
|
||||
static const void * say_api_buf[40];
|
||||
static const char *say_old = "old";
|
||||
static const char *say_new = "new";
|
||||
|
||||
static void save_say_mode(const void *arg)
|
||||
{
|
||||
int i = 0;
|
||||
say_api_buf[i++] = arg;
|
||||
|
||||
say_api_buf[i++] = ast_say_number_full;
|
||||
say_api_buf[i++] = ast_say_enumeration_full;
|
||||
say_api_buf[i++] = ast_say_digit_str_full;
|
||||
say_api_buf[i++] = ast_say_character_str_full;
|
||||
say_api_buf[i++] = ast_say_phonetic_str_full;
|
||||
say_api_buf[i++] = ast_say_datetime;
|
||||
say_api_buf[i++] = ast_say_time;
|
||||
say_api_buf[i++] = ast_say_date;
|
||||
say_api_buf[i++] = ast_say_datetime_from_now;
|
||||
say_api_buf[i++] = ast_say_date_with_format;
|
||||
}
|
||||
|
||||
static void restore_say_mode(void *arg)
|
||||
{
|
||||
int i = 0;
|
||||
say_api_buf[i++] = arg;
|
||||
|
||||
ast_say_number_full = say_api_buf[i++];
|
||||
ast_say_enumeration_full = say_api_buf[i++];
|
||||
ast_say_digit_str_full = say_api_buf[i++];
|
||||
ast_say_character_str_full = say_api_buf[i++];
|
||||
ast_say_phonetic_str_full = say_api_buf[i++];
|
||||
ast_say_datetime = say_api_buf[i++];
|
||||
ast_say_time = say_api_buf[i++];
|
||||
ast_say_date = say_api_buf[i++];
|
||||
ast_say_datetime_from_now = say_api_buf[i++];
|
||||
ast_say_date_with_format = say_api_buf[i++];
|
||||
}
|
||||
|
||||
/*
|
||||
* Typical 'say' arguments in addition to the date or number or string
|
||||
* to say. We do not include 'options' because they may be different
|
||||
* in recursive calls, and so they are better left as an external
|
||||
* parameter.
|
||||
*/
|
||||
typedef struct {
|
||||
struct ast_channel *chan;
|
||||
const char *ints;
|
||||
const char *language;
|
||||
int audiofd;
|
||||
int ctrlfd;
|
||||
} say_args_t;
|
||||
|
||||
static int s_streamwait3(const say_args_t *a, const char *fn)
|
||||
{
|
||||
int res = ast_streamfile(a->chan, fn, a->language);
|
||||
if (res) {
|
||||
ast_log(LOG_WARNING, "Unable to play message %s\n", fn);
|
||||
return res;
|
||||
}
|
||||
res = (a->audiofd > -1 && a->ctrlfd > -1) ?
|
||||
ast_waitstream_full(a->chan, a->ints, a->audiofd, a->ctrlfd) :
|
||||
ast_waitstream(a->chan, a->ints);
|
||||
ast_stopstream(a->chan);
|
||||
return res;
|
||||
}
|
||||
|
||||
/*
|
||||
* the string is 'prefix:data' or prefix:fmt:data'
|
||||
* with ':' being invalid in strings.
|
||||
*/
|
||||
static int do_say(say_args_t *a, const char *s, const char *options, int depth)
|
||||
{
|
||||
struct ast_variable *v;
|
||||
char *lang, *x, *rule = NULL;
|
||||
int ret = 0;
|
||||
struct varshead head = { .first = NULL, .last = NULL };
|
||||
struct ast_var_t *n;
|
||||
|
||||
ast_log(LOG_WARNING, "string <%s> depth <%d>\n", s, depth);
|
||||
if (depth++ > 10) {
|
||||
ast_log(LOG_WARNING, "recursion too deep, exiting\n");
|
||||
return -1;
|
||||
} else if (!say_cfg) {
|
||||
ast_log(LOG_WARNING, "no say.conf, cannot spell '%s'\n", s);
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* scan languages same as in file.c */
|
||||
if (a->language == NULL)
|
||||
a->language = "en"; /* default */
|
||||
ast_log(LOG_WARNING, "try <%s> in <%s>\n", s, a->language);
|
||||
lang = ast_strdupa(a->language);
|
||||
for (;;) {
|
||||
for (v = ast_variable_browse(say_cfg, lang); v ; v = v->next) {
|
||||
if (ast_extension_match(v->name, s)) {
|
||||
rule = ast_strdupa(v->value);
|
||||
break;
|
||||
}
|
||||
}
|
||||
if (rule)
|
||||
break;
|
||||
if ( (x = strchr(lang, '_')) )
|
||||
*x = '\0'; /* try without suffix */
|
||||
else if (strcmp(lang, "en"))
|
||||
lang = "en"; /* last resort, try 'en' if not done yet */
|
||||
else
|
||||
break;
|
||||
}
|
||||
if (!rule)
|
||||
return 0;
|
||||
|
||||
/* skip up to two prefixes to get the value */
|
||||
if ( (x = strchr(s, ':')) )
|
||||
s = x + 1;
|
||||
if ( (x = strchr(s, ':')) )
|
||||
s = x + 1;
|
||||
ast_log(LOG_WARNING, "value is <%s>\n", s);
|
||||
n = ast_var_assign("SAY", s);
|
||||
AST_LIST_INSERT_HEAD(&head, n, entries);
|
||||
|
||||
/* scan the body, one piece at a time */
|
||||
while ( ret <= 0 && (x = strsep(&rule, ",")) ) { /* exit on key */
|
||||
char fn[128];
|
||||
const char *p, *fmt, *data; /* format and data pointers */
|
||||
|
||||
/* prepare a decent file name */
|
||||
x = ast_skip_blanks(x);
|
||||
ast_trim_blanks(x);
|
||||
|
||||
/* replace variables */
|
||||
memset(fn, 0, sizeof(fn)); /* XXX why isn't done in pbx_substitute_variables_helper! */
|
||||
pbx_substitute_variables_varshead(&head, x, fn, sizeof(fn));
|
||||
ast_log(LOG_WARNING, "doing [%s]\n", fn);
|
||||
|
||||
/* locate prefix and data, if any */
|
||||
fmt = index(fn, ':');
|
||||
if (!fmt || fmt == fn) { /* regular filename */
|
||||
ret = s_streamwait3(a, fn);
|
||||
continue;
|
||||
}
|
||||
fmt++;
|
||||
data = index(fmt, ':'); /* colon before data */
|
||||
if (!data || data == fmt) { /* simple prefix-fmt */
|
||||
ret = do_say(a, fn, options, depth);
|
||||
continue;
|
||||
}
|
||||
/* prefix:fmt:data */
|
||||
for (p = fmt; p < data && ret <= 0; p++) {
|
||||
char fn2[sizeof(fn)];
|
||||
if (*p == ' ' || *p == '\t') /* skip blanks */
|
||||
continue;
|
||||
if (*p == '\'') {/* file name - we trim them */
|
||||
char *y;
|
||||
strcpy(fn2, ast_skip_blanks(p+1)); /* make a full copy */
|
||||
y = index(fn2, '\'');
|
||||
if (!y) {
|
||||
p = data; /* invalid. prepare to end */
|
||||
break;
|
||||
}
|
||||
*y = '\0';
|
||||
ast_trim_blanks(fn2);
|
||||
p = index(p+1, '\'');
|
||||
ret = s_streamwait3(a, fn2);
|
||||
} else {
|
||||
int l = fmt-fn;
|
||||
strcpy(fn2, fn); /* copy everything */
|
||||
/* after prefix, append the format */
|
||||
fn2[l++] = *p;
|
||||
strcpy(fn2 + l, data);
|
||||
ret = do_say(a, fn2, options, depth);
|
||||
}
|
||||
}
|
||||
}
|
||||
ast_var_delete(n);
|
||||
return ret;
|
||||
}
|
||||
|
||||
static int say_full(struct ast_channel *chan, const char *string,
|
||||
const char *ints, const char *lang, const char *options,
|
||||
int audiofd, int ctrlfd)
|
||||
{
|
||||
say_args_t a = { chan, ints, lang, audiofd, ctrlfd };
|
||||
return do_say(&a, string, options, 0);
|
||||
}
|
||||
|
||||
static int say_number_full(struct ast_channel *chan, int num,
|
||||
const char *ints, const char *lang, const char *options,
|
||||
int audiofd, int ctrlfd)
|
||||
{
|
||||
char buf[64];
|
||||
say_args_t a = { chan, ints, lang, audiofd, ctrlfd };
|
||||
snprintf(buf, sizeof(buf), "num:%d", num);
|
||||
return do_say(&a, buf, options, 0);
|
||||
}
|
||||
|
||||
static int say_enumeration_full(struct ast_channel *chan, int num,
|
||||
const char *ints, const char *lang, const char *options,
|
||||
int audiofd, int ctrlfd)
|
||||
{
|
||||
char buf[64];
|
||||
say_args_t a = { chan, ints, lang, audiofd, ctrlfd };
|
||||
snprintf(buf, sizeof(buf), "enum:%d", num);
|
||||
return do_say(&a, buf, options, 0);
|
||||
}
|
||||
|
||||
static int say_date_generic(struct ast_channel *chan, time_t t,
|
||||
const char *ints, const char *lang, const char *format, const char *timezone, const char *prefix)
|
||||
{
|
||||
char buf[128];
|
||||
struct tm tm;
|
||||
say_args_t a = { chan, ints, lang, -1, -1 };
|
||||
if (format == NULL)
|
||||
format = "";
|
||||
|
||||
ast_localtime(&t, &tm, NULL);
|
||||
snprintf(buf, sizeof(buf), "%s:%s:%04d%02d%02d%02d%02d.%02d-%d-%3d",
|
||||
prefix,
|
||||
format,
|
||||
tm.tm_year+1900,
|
||||
tm.tm_mon+1,
|
||||
tm.tm_mday,
|
||||
tm.tm_hour,
|
||||
tm.tm_min,
|
||||
tm.tm_sec,
|
||||
tm.tm_wday,
|
||||
tm.tm_yday);
|
||||
return do_say(&a, buf, NULL, 0);
|
||||
}
|
||||
|
||||
static int say_date_with_format(struct ast_channel *chan, time_t t,
|
||||
const char *ints, const char *lang, const char *format, const char *timezone)
|
||||
{
|
||||
return say_date_generic(chan, t, ints, lang, format, timezone, "datetime");
|
||||
}
|
||||
|
||||
static int say_date(struct ast_channel *chan, time_t t, const char *ints, const char *lang)
|
||||
{
|
||||
return say_date_generic(chan, t, ints, lang, "", NULL, "date");
|
||||
}
|
||||
|
||||
static int say_time(struct ast_channel *chan, time_t t, const char *ints, const char *lang)
|
||||
{
|
||||
return say_date_generic(chan, t, ints, lang, "", NULL, "time");
|
||||
}
|
||||
|
||||
static int say_datetime(struct ast_channel *chan, time_t t, const char *ints, const char *lang)
|
||||
{
|
||||
return say_date_generic(chan, t, ints, lang, "", NULL, "datetime");
|
||||
}
|
||||
|
||||
/*
|
||||
* remap the 'say' functions to use those in this file
|
||||
*/
|
||||
static int __say_init(int fd, int argc, char *argv[])
|
||||
{
|
||||
const char *old_mode = say_api_buf[0] ? say_new : say_old;
|
||||
char *mode;
|
||||
|
||||
if (argc == 2) {
|
||||
ast_cli(fd, "say mode is [%s]\n", old_mode);
|
||||
return RESULT_SUCCESS;
|
||||
} else if (argc != 3)
|
||||
return RESULT_SHOWUSAGE;
|
||||
mode = argv[2];
|
||||
|
||||
ast_log(LOG_WARNING, "init say.c from %s to %s\n", old_mode, mode);
|
||||
|
||||
if (!strcmp(mode, old_mode)) {
|
||||
ast_log(LOG_WARNING, "say mode is %s already\n", mode);
|
||||
} else if (!strcmp(mode, say_new)) {
|
||||
if (say_cfg == NULL)
|
||||
say_cfg = ast_config_load("say.conf");
|
||||
save_say_mode(say_new);
|
||||
ast_say_number_full = say_number_full;
|
||||
|
||||
ast_say_enumeration_full = say_enumeration_full;
|
||||
#if 0
|
||||
ast_say_digits_full = say_digits_full;
|
||||
ast_say_digit_str_full = say_digit_str_full;
|
||||
ast_say_character_str_full = say_character_str_full;
|
||||
ast_say_phonetic_str_full = say_phonetic_str_full;
|
||||
ast_say_datetime_from_now = say_datetime_from_now;
|
||||
#endif
|
||||
ast_say_datetime = say_datetime;
|
||||
ast_say_time = say_time;
|
||||
ast_say_date = say_date;
|
||||
ast_say_date_with_format = say_date_with_format;
|
||||
} else if (!strcmp(mode, say_old) && say_api_buf[0] == say_new) {
|
||||
restore_say_mode(NULL);
|
||||
} else {
|
||||
ast_log(LOG_WARNING, "unrecognized mode %s\n", mode);
|
||||
}
|
||||
return RESULT_SUCCESS;
|
||||
}
|
||||
|
||||
static struct ast_cli_entry cli_playback[] = {
|
||||
{ { "say", "load", NULL },
|
||||
__say_init, "set/show the say mode",
|
||||
"say load new|old" },
|
||||
};
|
||||
LOCAL_USER_DECL;
|
||||
|
||||
static int playback_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
int mres = 0;
|
||||
struct ast_module_user *u;
|
||||
char *tmp;
|
||||
struct localuser *u;
|
||||
char tmp[256];
|
||||
char *options;
|
||||
int option_skip=0;
|
||||
int option_say=0;
|
||||
int option_noanswer = 0;
|
||||
int priority_jump = 0;
|
||||
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(filenames);
|
||||
AST_APP_ARG(options);
|
||||
);
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
if (!data || !strlen((char *)data)) {
|
||||
ast_log(LOG_WARNING, "Playback requires an argument (filename)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
tmp = ast_strdupa(data);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
AST_STANDARD_APP_ARGS(args, tmp);
|
||||
|
||||
if (args.options) {
|
||||
if (strcasestr(args.options, "skip"))
|
||||
option_skip = 1;
|
||||
if (strcasestr(args.options, "say"))
|
||||
option_say = 1;
|
||||
if (strcasestr(args.options, "noanswer"))
|
||||
option_noanswer = 1;
|
||||
if (strchr(args.options, 'j'))
|
||||
priority_jump = 1;
|
||||
}
|
||||
|
||||
if (chan->_state != AST_STATE_UP) {
|
||||
strncpy(tmp, (char *)data, sizeof(tmp)-1);
|
||||
strtok(tmp, "|");
|
||||
options = strtok(NULL, "|");
|
||||
if (options && !strcasecmp(options, "skip"))
|
||||
option_skip = 1;
|
||||
if (options && !strcasecmp(options, "noanswer"))
|
||||
option_noanswer = 1;
|
||||
LOCAL_USER_ADD(u);
|
||||
if (chan->state != AST_STATE_UP) {
|
||||
if (option_skip) {
|
||||
/* At the user's option, skip if the line is not up */
|
||||
goto done;
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return 0;
|
||||
} else if (!option_noanswer)
|
||||
/* Otherwise answer unless we're supposed to send this while on-hook */
|
||||
res = ast_answer(chan);
|
||||
}
|
||||
if (!res) {
|
||||
char *back = args.filenames;
|
||||
char *front;
|
||||
|
||||
ast_stopstream(chan);
|
||||
while (!res && (front = strsep(&back, "&"))) {
|
||||
if (option_say)
|
||||
res = say_full(chan, front, "", chan->language, NULL, -1, -1);
|
||||
else
|
||||
res = ast_streamfile(chan, front, chan->language);
|
||||
if (!res) {
|
||||
res = ast_waitstream(chan, "");
|
||||
ast_stopstream(chan);
|
||||
} else {
|
||||
ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
|
||||
if (priority_jump || ast_opt_priority_jumping)
|
||||
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
|
||||
res = 0;
|
||||
mres = 1;
|
||||
}
|
||||
}
|
||||
res = ast_streamfile(chan, tmp, chan->language);
|
||||
if (!res)
|
||||
res = ast_waitstream(chan, "");
|
||||
else
|
||||
ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
|
||||
ast_stopstream(chan);
|
||||
}
|
||||
done:
|
||||
pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", mres ? "FAILED" : "SUCCESS");
|
||||
ast_module_user_remove(u);
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int reload(void)
|
||||
int unload_module(void)
|
||||
{
|
||||
if (say_cfg) {
|
||||
ast_config_destroy(say_cfg);
|
||||
ast_log(LOG_NOTICE, "Reloading say.conf\n");
|
||||
}
|
||||
say_cfg = ast_config_load("say.conf");
|
||||
/*
|
||||
* XXX here we should sort rules according to the same order
|
||||
* we have in pbx.c so we have the same matching behaviour.
|
||||
*/
|
||||
return 0;
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
int load_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_cli_unregister_multiple(cli_playback, sizeof(cli_playback) / sizeof(struct ast_cli_entry));
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
if (say_cfg)
|
||||
ast_config_destroy(say_cfg);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
say_cfg = ast_config_load("say.conf");
|
||||
ast_cli_register_multiple(cli_playback, sizeof(cli_playback) / sizeof(struct ast_cli_entry));
|
||||
return ast_register_application(app, playback_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Sound File Playback Application",
|
||||
.load = load_module,
|
||||
.unload = unload_module,
|
||||
.reload = reload,
|
||||
);
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
|
||||
char *key()
|
||||
{
|
||||
return ASTERISK_GPL_KEY;
|
||||
}
|
||||
|
||||
@@ -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 Block all calls without Caller*ID, require phone # to be entered
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
|
||||
#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 *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"
|
||||
;
|
||||
|
||||
|
||||
static int privacy_exec (struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res=0;
|
||||
int retries;
|
||||
int maxretries = 3;
|
||||
int minlength = 10;
|
||||
int x = 0;
|
||||
const char *s;
|
||||
char phone[30];
|
||||
struct ast_module_user *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);
|
||||
);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
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) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
|
||||
if (!ast_strlen_zero(data)) {
|
||||
parse = ast_strdupa(data);
|
||||
|
||||
AST_STANDARD_APP_ARGS(args, parse);
|
||||
|
||||
if (args.maxretries) {
|
||||
if (sscanf(args.maxretries, "%d", &x) == 1)
|
||||
maxretries = x;
|
||||
else
|
||||
ast_log(LOG_WARNING, "Invalid max retries argument\n");
|
||||
}
|
||||
if (args.minlength) {
|
||||
if (sscanf(args.minlength, "%d", &x) == 1)
|
||||
minlength = x;
|
||||
else
|
||||
ast_log(LOG_WARNING, "Invalid min length argument\n");
|
||||
}
|
||||
if (args.options)
|
||||
if (strchr(args.options, 'j'))
|
||||
priority_jump = 1;
|
||||
|
||||
}
|
||||
|
||||
if (!x)
|
||||
{
|
||||
/*Read in the config file*/
|
||||
cfg = ast_config_load(PRIV_CONFIG);
|
||||
|
||||
if (cfg && (s = ast_variable_retrieve(cfg, "general", "maxretries"))) {
|
||||
if (sscanf(s, "%d", &x) == 1)
|
||||
maxretries = x;
|
||||
else
|
||||
ast_log(LOG_WARNING, "Invalid max retries argument\n");
|
||||
}
|
||||
|
||||
if (cfg && (s = ast_variable_retrieve(cfg, "general", "minlength"))) {
|
||||
if (sscanf(s, "%d", &x) == 1)
|
||||
minlength = x;
|
||||
else
|
||||
ast_log(LOG_WARNING, "Invalid min length argument\n");
|
||||
}
|
||||
}
|
||||
|
||||
/*Play unidentified call*/
|
||||
res = ast_safe_sleep(chan, 1000);
|
||||
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 || ast_opt_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);
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application (app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application (app, privacy_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Require phone number to be entered, if no CallerID sent");
|
||||
370
apps/app_qcall.c
Normal file
370
apps/app_qcall.c
Normal file
@@ -0,0 +1,370 @@
|
||||
/** @file app_qcall.c
|
||||
*
|
||||
* Asterisk -- A telephony toolkit for Linux.
|
||||
*
|
||||
* Call back a party and connect them to a running pbx thread
|
||||
*
|
||||
* Copyright (C) 1999, Mark Spencer
|
||||
*
|
||||
* Mark Spencer <markster@linux-support.net>
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License
|
||||
*
|
||||
* Call a user from a file contained within a queue (/var/spool/asterisk/qcall)
|
||||
*
|
||||
* The queue is a directory containing files with the call request information
|
||||
* as a single line of text as follows:
|
||||
*
|
||||
* Dialstring Caller-ID Extension Maxsecs Identifier [Required-response]
|
||||
*
|
||||
* Dialstring -- A Dial String (The number to be called) in the
|
||||
* format Technology/Number, such IAX/mysys/1234 or Zap/g1/1234
|
||||
*
|
||||
* Caller-ID -- A Standard nomalized representation of the Caller-ID of
|
||||
* the number being dialed (generally 10 digits in the US).
|
||||
*
|
||||
* Extension -- The Extension (optionally Extension@context) that the
|
||||
* user should be "transferred" to after acceptance of the call.
|
||||
*
|
||||
* Maxsecs -- The Maximum time of the call in seconds. Specify 0 for infinite.
|
||||
*
|
||||
* Identifier -- The "Identifier" of the request. This is used to determine
|
||||
* the names of the audio prompt files played. The first prompt, the one that
|
||||
* asks for the input, is just the exact string specified as the identifier.
|
||||
* The second prompt, the one that is played after the correct input is given,
|
||||
* (generally a "thank you" recording), is the specified string with "-ok"
|
||||
* added to the end. So, if you specify "foo" as the identifier, your first
|
||||
* prompt file that will be played will be "foo" and the second one will be
|
||||
* "foo-ok".
|
||||
*
|
||||
* Required-Response (Optional) -- Specify a digit string to be used as the
|
||||
* acceptance "code" if you desire it to be something other then "1". This
|
||||
* can be used to implement some sort of PIN or security system. It may be
|
||||
* more then a single character.
|
||||
*
|
||||
* NOTE: It is important to remember that the process that creates these
|
||||
* files needs keep and maintain a write lock (using flock with the LOCK_EX
|
||||
* option) when writing these files.
|
||||
*
|
||||
*/
|
||||
|
||||
#include <asterisk/file.h>
|
||||
#include <asterisk/logger.h>
|
||||
#include <asterisk/channel.h>
|
||||
#include <asterisk/pbx.h>
|
||||
#include <asterisk/module.h>
|
||||
#include <asterisk/translate.h>
|
||||
#include <asterisk/options.h>
|
||||
#include <stdio.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
#include <stdlib.h>
|
||||
#include <pthread.h>
|
||||
#include <sys/types.h>
|
||||
#include <sys/stat.h>
|
||||
#include <errno.h>
|
||||
#include <dirent.h>
|
||||
#include <ctype.h>
|
||||
#include <sys/stat.h>
|
||||
#include <sys/time.h>
|
||||
#include <sys/file.h>
|
||||
|
||||
const char *qdir="/var/spool/asterisk/qcall";
|
||||
static char *tdesc = "Call from Queue";
|
||||
static pthread_t qcall_thread;
|
||||
static int debug = 0;
|
||||
STANDARD_LOCAL_USER;
|
||||
LOCAL_USER_DECL;
|
||||
|
||||
#define OLDESTOK 14400 /* not any more then this number of secs old */
|
||||
#define INITIALONE 20 /* initial wait before the first one in secs */
|
||||
#define NEXTONE 600 /* wait before trying it again in secs */
|
||||
#define MAXWAITFORANSWER 45000 /* max call time before answer */
|
||||
/* define either one or both of these two if your application requires it */
|
||||
#if 0
|
||||
#define ACCTCODE "SOMETHING" /* Account code */
|
||||
#define AMAFLAGS AST_CDR_BILLING /* AMA flags */
|
||||
#endif
|
||||
/* define this if you want to have a particular CLID display on the user's
|
||||
phone when they receive the call */
|
||||
#if 0
|
||||
#define OURCLID "2564286275" /* The callerid to be displayed when calling */
|
||||
#endif
|
||||
|
||||
static void *qcall_do(void *arg);
|
||||
|
||||
static void *qcall(void *ignore)
|
||||
{
|
||||
pthread_t dialer_thread;
|
||||
DIR *dirp;
|
||||
FILE *fp;
|
||||
struct dirent *dp;
|
||||
char fname[80];
|
||||
struct stat mystat;
|
||||
time_t t;
|
||||
void *arg;
|
||||
pthread_attr_t attr;
|
||||
|
||||
time(&t);
|
||||
if (debug) printf("@@@@ qcall starting at %s",ctime(&t));
|
||||
for(;;)
|
||||
{
|
||||
time(&t);
|
||||
dirp = opendir(qdir);
|
||||
if (!dirp)
|
||||
{
|
||||
perror("app_qcall:Cannot open queue directory");
|
||||
break;
|
||||
}
|
||||
while((dp = readdir(dirp)) != NULL)
|
||||
{
|
||||
if (dp->d_name[0] == '.') continue;
|
||||
sprintf(fname,"%s/%s",qdir,dp->d_name);
|
||||
if (stat(fname,&mystat) == -1)
|
||||
{
|
||||
perror("app_qcall:stat");
|
||||
fprintf(stderr,"%s\n",fname);
|
||||
continue;
|
||||
}
|
||||
/* if not a regular file, skip it */
|
||||
if ((mystat.st_mode & S_IFMT) != S_IFREG) continue;
|
||||
/* if not yet .... */
|
||||
if (mystat.st_atime == mystat.st_ctime)
|
||||
{ /* first time */
|
||||
if ((mystat.st_atime + INITIALONE) > t) continue;
|
||||
}
|
||||
else
|
||||
{ /* already looked at once */
|
||||
if ((mystat.st_atime + NEXTONE) > t) continue;
|
||||
}
|
||||
/* if too old */
|
||||
if (mystat.st_mtime < (t - OLDESTOK))
|
||||
{
|
||||
/* kill it, its too old */
|
||||
unlink(fname);
|
||||
continue;
|
||||
}
|
||||
/* "touch" file's access time */
|
||||
fp = fopen(fname,"r");
|
||||
if (fp) fclose(fp);
|
||||
/* make a copy of the filename string, so that we
|
||||
may go on and use the buffer */
|
||||
arg = (void *) strdup(fname);
|
||||
pthread_attr_init(&attr);
|
||||
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
|
||||
if (pthread_create(&dialer_thread,&attr,qcall_do,arg) == -1)
|
||||
{
|
||||
perror("qcall: Cannot create thread");
|
||||
continue;
|
||||
}
|
||||
}
|
||||
closedir(dirp);
|
||||
sleep(1);
|
||||
}
|
||||
pthread_exit(NULL);
|
||||
}
|
||||
|
||||
/* single thread with one file (request) to dial */
|
||||
static void *qcall_do(void *arg)
|
||||
{
|
||||
char fname[300],dialstr[300],extstr[300],ident[300],reqinp[300],buf[300];
|
||||
char clid[300],*tele,*context;
|
||||
FILE *fp;
|
||||
int ms = MAXWAITFORANSWER,maxsecs;
|
||||
struct ast_channel *channel;
|
||||
time_t t;
|
||||
|
||||
/* get the filename from the arg */
|
||||
strcpy(fname,(char *)arg);
|
||||
free(arg);
|
||||
time(&t);
|
||||
fp = fopen(fname,"r");
|
||||
if (!fp) /* if cannot open request file */
|
||||
{
|
||||
perror("qcall_do:fopen");
|
||||
fprintf(stderr,"%s\n",fname);
|
||||
unlink(fname);
|
||||
pthread_exit(NULL);
|
||||
}
|
||||
/* lock the file */
|
||||
if (flock(fileno(fp),LOCK_EX) == -1)
|
||||
{
|
||||
perror("qcall_do:flock");
|
||||
fprintf(stderr,"%s\n",fname);
|
||||
pthread_exit(NULL);
|
||||
}
|
||||
strcpy(reqinp,"1"); /* default required input for acknowledgement */
|
||||
if (fscanf(fp,"%s %s %s %d %s %s",dialstr,clid,
|
||||
extstr,&maxsecs,ident,reqinp) < 5)
|
||||
{
|
||||
fprintf(stderr,"qcall_do:file line invalid in file %s:\n",fname);
|
||||
pthread_exit(NULL);
|
||||
}
|
||||
flock(fileno(fp),LOCK_UN);
|
||||
fclose(fp);
|
||||
tele = strchr(dialstr,'/');
|
||||
if (!tele)
|
||||
{
|
||||
fprintf(stderr,"qcall_do:Dial number must be in format tech/number\n");
|
||||
unlink(fname);
|
||||
pthread_exit(NULL);
|
||||
}
|
||||
*tele++ = 0;
|
||||
channel = ast_request(dialstr,AST_FORMAT_SLINEAR,tele);
|
||||
if (channel)
|
||||
{
|
||||
ast_set_read_format(channel,AST_FORMAT_SLINEAR);
|
||||
ast_set_write_format(channel,AST_FORMAT_SLINEAR);
|
||||
channel->callerid = NULL;
|
||||
channel->ani = NULL;
|
||||
#ifdef OURCLID
|
||||
channel->callerid = strdup(OURCLID);
|
||||
channel->ani = strdup(OURCLID);
|
||||
#endif
|
||||
channel->whentohangup = 0;
|
||||
channel->appl = "AppQcall";
|
||||
channel->data = "(Outgoing Line)";
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "Qcall initiating call to %s/%s on %s (%s)\n",
|
||||
dialstr,tele,channel->name,fname);
|
||||
ast_call(channel,tele,MAXWAITFORANSWER);
|
||||
}
|
||||
else
|
||||
{
|
||||
fprintf(stderr,"qcall_do:Sorry unable to obtain channel\n");
|
||||
pthread_exit(NULL);
|
||||
}
|
||||
if (channel->callerid) free(channel->callerid);
|
||||
channel->callerid = NULL;
|
||||
if (channel->ani) free(channel->ani);
|
||||
channel->ani = NULL;
|
||||
if (channel->state == AST_STATE_UP)
|
||||
if (debug) printf("@@@@ Autodial:Line is Up\n");
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "Qcall waiting for answer on %s\n",
|
||||
channel->name);
|
||||
while(ms > 0){
|
||||
struct ast_frame *f;
|
||||
ms = ast_waitfor(channel,ms);
|
||||
f = ast_read(channel);
|
||||
if (!f)
|
||||
{
|
||||
if (debug) printf("@@@@ qcall_do:Hung Up\n");
|
||||
ast_frfree(f);
|
||||
unlink(fname);
|
||||
break;
|
||||
}
|
||||
if (f->frametype == AST_FRAME_CONTROL)
|
||||
{
|
||||
if (f->subclass == AST_CONTROL_HANGUP)
|
||||
{
|
||||
if (debug) printf("@@@@ qcall_do:Hung Up\n");
|
||||
unlink(fname);
|
||||
ast_frfree(f);
|
||||
break;
|
||||
}
|
||||
if (f->subclass == AST_CONTROL_ANSWER)
|
||||
{
|
||||
if (debug) printf("@@@@ qcall_do:Phone Answered\n");
|
||||
if (channel->state == AST_STATE_UP)
|
||||
{
|
||||
unlink(fname);
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "Qcall got answer on %s\n",
|
||||
channel->name);
|
||||
usleep(1500000);
|
||||
ast_streamfile(channel,ident,0);
|
||||
if (ast_readstring(channel,buf,strlen(reqinp),10000,5000,"#"))
|
||||
{
|
||||
ast_stopstream(channel);
|
||||
if (debug) printf("@@@@ qcall_do: timeout or hangup in dtmf read\n");
|
||||
ast_frfree(f);
|
||||
break;
|
||||
}
|
||||
ast_stopstream(channel);
|
||||
if (strcmp(buf,reqinp)) /* if not match */
|
||||
{
|
||||
if (debug) printf("@@@@ qcall_do: response (%s) does not match required (%s)\n",buf,reqinp);
|
||||
ast_frfree(f);
|
||||
break;
|
||||
}
|
||||
ast_frfree(f);
|
||||
/* okay, now we go for it */
|
||||
context = strchr(extstr,'@');
|
||||
if (!context) context = "default";
|
||||
else *context++ = 0;
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "Qcall got accept, now putting through to %s@%s on %s\n",
|
||||
extstr,context,channel->name);
|
||||
strcat(ident,"-ok");
|
||||
/* if file existant, play it */
|
||||
if (!ast_streamfile(channel,ident,0))
|
||||
{
|
||||
ast_waitstream(channel,"");
|
||||
ast_stopstream(channel);
|
||||
}
|
||||
channel->callerid = strdup(clid);
|
||||
channel->ani = strdup(clid);
|
||||
channel->language[0] = 0;
|
||||
channel->dnid = strdup(extstr);
|
||||
#ifdef AMAFLAGS
|
||||
channel->amaflags = AMAFLAGS;
|
||||
#endif
|
||||
#ifdef ACCTCODE
|
||||
strcpy(channel->accountcode,ACCTCODE);
|
||||
#else
|
||||
channel->accountcode[0] = 0;
|
||||
#endif
|
||||
if (maxsecs) /* if finite length call */
|
||||
{
|
||||
time(&channel->whentohangup);
|
||||
channel->whentohangup += maxsecs;
|
||||
}
|
||||
strcpy(channel->exten,extstr);
|
||||
strcpy(channel->context,context);
|
||||
channel->priority = 1;
|
||||
ast_pbx_run(channel);
|
||||
pthread_exit(NULL);
|
||||
}
|
||||
}
|
||||
else if(f->subclass==AST_CONTROL_RINGING)
|
||||
if (debug) printf("@@@@ qcall_do:Phone Ringing end\n");
|
||||
}
|
||||
ast_frfree(f);
|
||||
}
|
||||
ast_hangup(channel);
|
||||
if (debug) printf("@@@@ qcall_do:Hung up channel\n");
|
||||
pthread_exit(NULL);
|
||||
return NULL;
|
||||
}
|
||||
|
||||
int unload_module(void)
|
||||
{
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
return 0;
|
||||
}
|
||||
|
||||
int load_module(void)
|
||||
{
|
||||
mkdir(qdir,0660);
|
||||
pthread_create(&qcall_thread,NULL,qcall,NULL);
|
||||
return 0;
|
||||
}
|
||||
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
|
||||
char *key()
|
||||
{
|
||||
return ASTERISK_GPL_KEY;
|
||||
}
|
||||
4526
apps/app_queue.c
4526
apps/app_queue.c
File diff suppressed because it is too large
Load Diff
@@ -1,108 +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 "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
|
||||
/*! \todo The Random() app should be removed from trunk following the release of 1.4 */
|
||||
|
||||
static char *app_random = "Random";
|
||||
|
||||
static char *random_synopsis = "Conditionally branches, based upon a probability";
|
||||
|
||||
static char *random_descrip =
|
||||
"Random([probability]:[[context|]extension|]priority)\n"
|
||||
" probability := INTEGER in the range 1 to 100\n"
|
||||
"DEPRECATED: Use GotoIf($[${RAND(1,100)} > <number>]?<label>)\n";
|
||||
|
||||
|
||||
static int random_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res=0;
|
||||
struct ast_module_user *u;
|
||||
|
||||
char *s;
|
||||
char *prob;
|
||||
int probint;
|
||||
static int deprecated = 0;
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "Random requires an argument ([probability]:[[context|]extension|]priority)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
s = ast_strdupa(data);
|
||||
|
||||
prob = strsep(&s,":");
|
||||
if ((!prob) || (sscanf(prob, "%d", &probint) != 1))
|
||||
probint = 0;
|
||||
|
||||
if (!deprecated) {
|
||||
deprecated = 1;
|
||||
ast_log(LOG_WARNING, "Random is deprecated in Asterisk 1.4. Replace with GotoIf($[${RAND(0,99)} + %d >= 100]?%s)\n", probint, s);
|
||||
}
|
||||
|
||||
if ((ast_random() % 100) + probint >= 100) {
|
||||
res = ast_parseable_goto(chan, s);
|
||||
if (option_verbose > 2)
|
||||
ast_verbose( VERBOSE_PREFIX_3 "Random branches to (%s,%s,%d)\n",
|
||||
chan->context,chan->exten, chan->priority+1);
|
||||
}
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app_random);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app_random, random_exec, random_synopsis, random_descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Random goto");
|
||||
234
apps/app_read.c
234
apps/app_read.c
@@ -1,234 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 1999 - 2005, Digium, Inc.
|
||||
*
|
||||
* Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief Trivial application to read a variable
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/translate.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/utils.h"
|
||||
#include "asterisk/indications.h"
|
||||
|
||||
enum {
|
||||
OPT_SKIP = (1 << 0),
|
||||
OPT_INDICATION = (1 << 1),
|
||||
OPT_NOANSWER = (1 << 2),
|
||||
} read_option_flags;
|
||||
|
||||
AST_APP_OPTIONS(read_app_options, {
|
||||
AST_APP_OPTION('s', OPT_SKIP),
|
||||
AST_APP_OPTION('i', OPT_INDICATION),
|
||||
AST_APP_OPTION('n', OPT_NOANSWER),
|
||||
});
|
||||
|
||||
static char *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 or tone with option i\n"
|
||||
" maxdigits -- maximum acceptable number of digits. Stops reading after\n"
|
||||
" maxdigits have been entered (without requiring the user to\n"
|
||||
" press the '#' key).\n"
|
||||
" Defaults to 0 - no limit - wait for the user press the '#' key.\n"
|
||||
" Any value below 0 means the same. Max accepted value is 255.\n"
|
||||
" option -- options are 's' , 'i', 'n'\n"
|
||||
" 's' to return immediately if the line is not up,\n"
|
||||
" 'i' to play filename as an indication tone from your indications.conf\n"
|
||||
" 'n' to read digits even if the line is not up.\n"
|
||||
" attempts -- if greater than 1, that many attempts will be made in the \n"
|
||||
" event no data is entered.\n"
|
||||
" timeout -- An integer number of seconds to wait for a digit response. If greater\n"
|
||||
" than 0, that value will override the default timeout.\n\n"
|
||||
"Read should disconnect if the function fails or errors out.\n";
|
||||
|
||||
|
||||
#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 ast_module_user *u;
|
||||
char tmp[256] = "";
|
||||
int maxdigits = 255;
|
||||
int tries = 1, to = 0, x = 0;
|
||||
char *argcopy = NULL;
|
||||
struct tone_zone_sound *ts;
|
||||
struct ast_flags flags = {0};
|
||||
|
||||
AST_DECLARE_APP_ARGS(arglist,
|
||||
AST_APP_ARG(variable);
|
||||
AST_APP_ARG(filename);
|
||||
AST_APP_ARG(maxdigits);
|
||||
AST_APP_ARG(options);
|
||||
AST_APP_ARG(attempts);
|
||||
AST_APP_ARG(timeout);
|
||||
);
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "Read requires an argument (variable)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
argcopy = ast_strdupa(data);
|
||||
|
||||
AST_STANDARD_APP_ARGS(arglist, argcopy);
|
||||
|
||||
if (!ast_strlen_zero(arglist.options)) {
|
||||
ast_app_parse_options(read_app_options, &flags, NULL, arglist.options);
|
||||
}
|
||||
|
||||
if (!ast_strlen_zero(arglist.attempts)) {
|
||||
tries = atoi(arglist.attempts);
|
||||
if (tries <= 0)
|
||||
tries = 1;
|
||||
}
|
||||
|
||||
if (!ast_strlen_zero(arglist.timeout)) {
|
||||
to = atoi(arglist.timeout);
|
||||
if (to <= 0)
|
||||
to = 0;
|
||||
else
|
||||
to *= 1000;
|
||||
}
|
||||
|
||||
if (ast_strlen_zero(arglist.filename)) {
|
||||
arglist.filename = NULL;
|
||||
}
|
||||
if (!ast_strlen_zero(arglist.maxdigits)) {
|
||||
maxdigits = atoi(arglist.maxdigits);
|
||||
if ((maxdigits<1) || (maxdigits>255)) {
|
||||
maxdigits = 255;
|
||||
} else if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "Accepting a maximum of %d digits.\n", maxdigits);
|
||||
}
|
||||
if (ast_strlen_zero(arglist.variable)) {
|
||||
ast_log(LOG_WARNING, "Invalid! Usage: Read(variable[|filename][|maxdigits][|option][|attempts][|timeout])\n\n");
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
ts=NULL;
|
||||
if (ast_test_flag(&flags,OPT_INDICATION)) {
|
||||
if (!ast_strlen_zero(arglist.filename)) {
|
||||
ts = ast_get_indication_tone(chan->zone,arglist.filename);
|
||||
}
|
||||
}
|
||||
if (chan->_state != AST_STATE_UP) {
|
||||
if (ast_test_flag(&flags,OPT_SKIP)) {
|
||||
/* At the user's option, skip if the line is not up */
|
||||
pbx_builtin_setvar_helper(chan, arglist.variable, "\0");
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
} else if (!ast_test_flag(&flags,OPT_NOANSWER)) {
|
||||
/* Otherwise answer unless we're supposed to read while on-hook */
|
||||
res = ast_answer(chan);
|
||||
}
|
||||
}
|
||||
if (!res) {
|
||||
while (tries && !res) {
|
||||
ast_stopstream(chan);
|
||||
if (ts && ts->data[0]) {
|
||||
if (!to)
|
||||
to = chan->pbx ? chan->pbx->rtimeout * 1000 : 6000;
|
||||
res = ast_playtones_start(chan, 0, ts->data, 0);
|
||||
for (x = 0; x < maxdigits; ) {
|
||||
res = ast_waitfordigit(chan, to);
|
||||
ast_playtones_stop(chan);
|
||||
if (res < 1) {
|
||||
tmp[x]='\0';
|
||||
break;
|
||||
}
|
||||
tmp[x++] = res;
|
||||
if (tmp[x-1] == '#') {
|
||||
tmp[x-1] = '\0';
|
||||
break;
|
||||
}
|
||||
}
|
||||
} else {
|
||||
res = ast_app_getdata(chan, arglist.filename, tmp, maxdigits, to);
|
||||
}
|
||||
if (res > -1) {
|
||||
pbx_builtin_setvar_helper(chan, arglist.variable, 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 {
|
||||
pbx_builtin_setvar_helper(chan, arglist.variable, tmp);
|
||||
if (option_verbose > 2)
|
||||
ast_verbose(VERBOSE_PREFIX_3 "User disconnected\n");
|
||||
}
|
||||
}
|
||||
}
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, read_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Read Variable Application");
|
||||
@@ -1,120 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 1999 - 2005, Digium, Inc.
|
||||
*
|
||||
* Matt O'Gorman <mogorman@digium.com>
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief ReadFile application -- Reads in a File for you.
|
||||
*
|
||||
* \author Matt O'Gorman <mogorman@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/module.h"
|
||||
|
||||
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";
|
||||
|
||||
|
||||
static int readfile_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res=0;
|
||||
struct ast_module_user *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;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
s = ast_strdupa(data);
|
||||
|
||||
varname = strsep(&s, "=");
|
||||
file = strsep(&s, "|");
|
||||
length = s;
|
||||
|
||||
if (!varname || !file) {
|
||||
ast_log(LOG_ERROR, "No file or variable specified!\n");
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
if (length) {
|
||||
if ((sscanf(length, "%d", &len) != 1) || (len < 0)) {
|
||||
ast_log(LOG_WARNING, "%s is not a positive number, defaulting length to max\n", length);
|
||||
len = 0;
|
||||
}
|
||||
}
|
||||
|
||||
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);
|
||||
}
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app_readfile);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app_readfile, readfile_exec, readfile_synopsis, readfile_descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Stores output of file into a variable");
|
||||
@@ -1,261 +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
|
||||
*
|
||||
* \author Anthony Minessale <anthmct@yahoo.com>
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/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 *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\n"
|
||||
"to the name. For example, a prefix of 'var_' would make the column 'name'\n"
|
||||
"become the variable ${var_name}. REALTIMECOUNT will be set with the number\n"
|
||||
"of values read.\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\n"
|
||||
"updated to <newval>. REALTIMECOUNT will be set with the number of rows\n"
|
||||
"updated or -1 if an error occurs.\n";
|
||||
|
||||
|
||||
static int cli_realtime_load(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_realtime_update(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_realtime_load_usage[] =
|
||||
"Usage: realtime load <family> <colmatch> <value>\n"
|
||||
" Prints out a list of variables using the RealTime driver.\n";
|
||||
|
||||
static char cli_realtime_update_usage[] =
|
||||
"Usage: realtime update <family> <colmatch> <value>\n"
|
||||
" Update a single variable using the RealTime driver.\n";
|
||||
|
||||
static struct ast_cli_entry cli_realtime[] = {
|
||||
{ { "realtime", "load", NULL, NULL },
|
||||
cli_realtime_load, "Used to print out RealTime variables.",
|
||||
cli_realtime_load_usage, NULL },
|
||||
|
||||
{ { "realtime", "update", NULL, NULL },
|
||||
cli_realtime_update, "Used to update RealTime variables.",
|
||||
cli_realtime_update_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 ast_module_user *u;
|
||||
int res = 0, count = 0;
|
||||
char countc[13];
|
||||
|
||||
ast_log(LOG_WARNING, "The RealTimeUpdate application has been deprecated in favor of the REALTIME dialplan function.\n");
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_ERROR,"Invalid input: usage %s\n",UUSAGE);
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
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 {
|
||||
count = ast_update_realtime(family,colmatch,value,newcol,newval,NULL);
|
||||
}
|
||||
|
||||
snprintf(countc, sizeof(countc), "%d", count);
|
||||
pbx_builtin_setvar_helper(chan, "REALTIMECOUNT", countc);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
|
||||
static int realtime_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res=0, count=0;
|
||||
struct ast_module_user *u;
|
||||
struct ast_variable *var, *itt;
|
||||
char *family=NULL, *colmatch=NULL, *value=NULL, *prefix=NULL, *vname=NULL;
|
||||
char countc[13];
|
||||
size_t len;
|
||||
|
||||
ast_log(LOG_WARNING, "The RealTime application has been deprecated in favor of the REALTIME dialplan function.\n");
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_ERROR,"Invalid input: usage %s\n",USAGE);
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
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);
|
||||
count++;
|
||||
}
|
||||
ast_variables_destroy(var);
|
||||
} else if (option_verbose > 3)
|
||||
ast_verbose(VERBOSE_PREFIX_4"No Realtime Matches Found.\n");
|
||||
}
|
||||
snprintf(countc, sizeof(countc), "%d", count);
|
||||
pbx_builtin_setvar_helper(chan, "REALTIMECOUNT", countc);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
ast_cli_unregister_multiple(cli_realtime, sizeof(cli_realtime) / sizeof(struct ast_cli_entry));
|
||||
res = ast_unregister_application(uapp);
|
||||
res |= ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
ast_cli_register_multiple(cli_realtime, sizeof(cli_realtime) / sizeof(struct ast_cli_entry));
|
||||
res = ast_register_application(uapp, realtime_update_exec, usynopsis, udesc);
|
||||
res |= ast_register_application(app, realtime_exec, synopsis, desc);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Realtime Data Lookup/Rewrite");
|
||||
@@ -1,386 +1,174 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
* Asterisk -- A telephony toolkit for Linux.
|
||||
*
|
||||
* Copyright (C) 1999 - 2005, Digium, Inc.
|
||||
* Trivial application to record a sound file
|
||||
*
|
||||
* Copyright (C) 2001, Linux Support Services, Inc.
|
||||
*
|
||||
* Matthew Fredrickson <creslin@digium.com>
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
* Matthew Fredrickson <creslin@linux-support.net>
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief Trivial application to record a sound file
|
||||
*
|
||||
* \author Matthew Fredrickson <creslin@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.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 <string.h>
|
||||
#include <stdlib.h>
|
||||
#include <pthread.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/translate.h"
|
||||
#include "asterisk/dsp.h"
|
||||
#include "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 (DTMF) instead of default '#'\n"
|
||||
" 'x' : ignore all terminator keys (DTMF) and keep recording until hangup\n"
|
||||
"\n"
|
||||
"If filename contains '%d', these characters will be replaced with a number\n"
|
||||
"incremented by one each time the file is recorded. A channel variable\n"
|
||||
"named RECORDED_FILE will also be set, which contains the final filemname.\n\n"
|
||||
"Use 'core show file formats' to see the available formats on your system\n\n"
|
||||
"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";
|
||||
" Record(filename:extension): Records from the channel into a given\n"
|
||||
"filename. If the file exists it will be overwritten. The 'extension'\n"
|
||||
"is the extension of the file type to be recorded (wav, gsm, etc).\n"
|
||||
"Returns -1 when the user hangs up.\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 fil[256];
|
||||
char tmp[256];
|
||||
|
||||
char ext[10];
|
||||
char * vdata; /* Used so I don't have to typecast every use of *data */
|
||||
int i = 0;
|
||||
int j = 0;
|
||||
|
||||
struct ast_filestream *s = '\0';
|
||||
struct ast_module_user *u;
|
||||
struct ast_frame *f = NULL;
|
||||
struct localuser *u;
|
||||
struct ast_frame *f;
|
||||
|
||||
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;
|
||||
vdata = data; /* explained above */
|
||||
|
||||
|
||||
/* The next few lines of code parse out the filename and header from the input string */
|
||||
if (ast_strlen_zero(data)) { /* no data implies no filename or anything is present */
|
||||
if (!vdata) { /* no data implies no filename or anything is present */
|
||||
ast_log(LOG_WARNING, "Record requires an argument (filename)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
/* Yay for strsep being easy */
|
||||
vdata = ast_strdupa(data);
|
||||
|
||||
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++;
|
||||
for (; vdata[i] && (vdata[i] != ':') ; i++ ) {
|
||||
if ((vdata[i] == '%') && (vdata[i+1] == 'd')) {
|
||||
percentflag = 1; /* the wildcard is used */
|
||||
}
|
||||
}
|
||||
if (!ext) {
|
||||
ast_log(LOG_WARNING, "No extension specified to filename!\n");
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
if (silstr) {
|
||||
if ((sscanf(silstr, "%d", &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 (i == strlen(vdata) ) {
|
||||
ast_log(LOG_WARNING, "No extension found\n");
|
||||
return -1;
|
||||
}
|
||||
fil[i] = vdata[i];
|
||||
}
|
||||
|
||||
if (maxstr) {
|
||||
if ((sscanf(maxstr, "%d", &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, 'x'))
|
||||
terminator = 0;
|
||||
if (strchr(options, 'q'))
|
||||
option_quiet = 1;
|
||||
}
|
||||
}
|
||||
|
||||
fil[i++] = '\0';
|
||||
|
||||
for (; j < 10 && i < strlen(data); i++, j++)
|
||||
ext[j] = vdata[i];
|
||||
ext[j] = '\0';
|
||||
/* 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, '%');
|
||||
do {
|
||||
int tmplen;
|
||||
/* First piece has no leading percent, so it's copied verbatim */
|
||||
ast_copy_string(tmp, fname.piece[0], sizeof(tmp));
|
||||
tmplen = strlen(tmp);
|
||||
for (i = 1; i < fname.argc; i++) {
|
||||
if (fname.piece[i][0] == 'd') {
|
||||
/* Substitute the count */
|
||||
snprintf(countstring, sizeof(countstring), "%d", count);
|
||||
ast_copy_string(tmp + tmplen, countstring, sizeof(tmp) - tmplen);
|
||||
tmplen += strlen(countstring);
|
||||
} else if (tmplen + 2 < sizeof(tmp)) {
|
||||
/* Unknown format specifier - just copy it verbatim */
|
||||
tmp[tmplen++] = '%';
|
||||
tmp[tmplen++] = fname.piece[i][0];
|
||||
}
|
||||
/* Copy the remaining portion of the piece */
|
||||
ast_copy_string(tmp + tmplen, &(fname.piece[i][1]), sizeof(tmp) - tmplen);
|
||||
}
|
||||
snprintf(tmp, 256, fil, count);
|
||||
count++;
|
||||
} while (ast_fileexists(tmp, ext, chan->language) > 0);
|
||||
pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
|
||||
} while ( ast_fileexists(tmp, ext, chan->language) != -1 );
|
||||
} else
|
||||
ast_copy_string(tmp, filename, sizeof(tmp));
|
||||
strncpy(tmp, fil, 256-1);
|
||||
/* 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 */
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
} else if (!option_noanswer) {
|
||||
/* Otherwise answer unless we're supposed to record while on-hook */
|
||||
res = ast_answer(chan);
|
||||
}
|
||||
|
||||
LOCAL_USER_ADD(u);
|
||||
|
||||
if (chan->state != AST_STATE_UP) {
|
||||
res = ast_answer(chan); /* Shouldn't need this, but checking to see if channel is already answered
|
||||
* Theoretically asterisk should already have answered before running the app */
|
||||
}
|
||||
|
||||
if (res) {
|
||||
ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
|
||||
goto out;
|
||||
}
|
||||
|
||||
if (!option_quiet) {
|
||||
if (!res) {
|
||||
/* Some code to play a nice little beep to signify the start of the record operation */
|
||||
res = ast_streamfile(chan, "beep", chan->language);
|
||||
if (!res) {
|
||||
printf("Waiting on stream\n");
|
||||
res = ast_waitstream(chan, "");
|
||||
} else {
|
||||
printf("streamfile failed\n");
|
||||
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");
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
sildet = ast_dsp_new();
|
||||
if (!sildet) {
|
||||
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
|
||||
ast_module_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 (ast_opt_transmit_silence)
|
||||
silgen = ast_channel_start_silence_generator(chan);
|
||||
/* The end of beep code. Now the recording starts */
|
||||
s = ast_writefile( tmp, ext, NULL, O_CREAT|O_TRUNC|O_WRONLY , 0, 0644);
|
||||
|
||||
/* Request a video update */
|
||||
ast_indicate(chan, AST_CONTROL_VIDUPDATE);
|
||||
|
||||
if (maxduration <= 0)
|
||||
maxduration = -1;
|
||||
|
||||
while ((waitres = ast_waitfor(chan, maxduration)) > -1) {
|
||||
if (maxduration > 0) {
|
||||
if (waitres == 0) {
|
||||
gottimeout = 1;
|
||||
break;
|
||||
}
|
||||
maxduration = waitres;
|
||||
}
|
||||
if (s) {
|
||||
|
||||
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;
|
||||
while ((f = ast_read(chan))) {
|
||||
if (f->frametype == AST_FRAME_VOICE) {
|
||||
res = ast_writestream(s, f);
|
||||
|
||||
if (res) {
|
||||
ast_log(LOG_WARNING, "Problem writing frame\n");
|
||||
break;
|
||||
}
|
||||
}
|
||||
if (totalsilence > silence) {
|
||||
/* Ended happily with silence */
|
||||
if ((f->frametype == AST_FRAME_DTMF) &&
|
||||
(f->subclass == '#')) {
|
||||
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);
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
if (!f) {
|
||||
ast_log(LOG_DEBUG, "Got hangup\n");
|
||||
res = -1;
|
||||
}
|
||||
ast_closestream(s);
|
||||
} else
|
||||
ast_log(LOG_WARNING, "Could not create file %s\n", fil);
|
||||
} else
|
||||
ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
|
||||
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, record_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Trivial Record Application");
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
|
||||
char *key()
|
||||
{
|
||||
return ASTERISK_GPL_KEY;
|
||||
}
|
||||
|
||||
8059
apps/app_rpt.c
8059
apps/app_rpt.c
File diff suppressed because it is too large
Load Diff
@@ -1,126 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (c) 2003, 2006 Tilghman Lesher. All rights reserved.
|
||||
* Copyright (c) 2006 Digium, Inc.
|
||||
*
|
||||
* Tilghman Lesher <app_sayunixtime__200309@the-tilghman.com>
|
||||
*
|
||||
* This code is released by the author with no restrictions on usage.
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief SayUnixTime application
|
||||
*
|
||||
* \author Tilghman Lesher <app_sayunixtime__200309@the-tilghman.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/say.h"
|
||||
#include "asterisk/app.h"
|
||||
|
||||
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";
|
||||
|
||||
|
||||
static int sayunixtime_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(timeval);
|
||||
AST_APP_ARG(timezone);
|
||||
AST_APP_ARG(format);
|
||||
);
|
||||
char *parse;
|
||||
int res = 0;
|
||||
struct ast_module_user *u;
|
||||
time_t unixtime;
|
||||
|
||||
if (!data)
|
||||
return 0;
|
||||
|
||||
parse = ast_strdupa(data);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
AST_STANDARD_APP_ARGS(args, parse);
|
||||
|
||||
ast_get_time_t(args.timeval, &unixtime, time(NULL), NULL);
|
||||
|
||||
if (chan->_state != AST_STATE_UP)
|
||||
res = ast_answer(chan);
|
||||
|
||||
if (!res)
|
||||
res = ast_say_date_with_format(chan, unixtime, AST_DIGIT_ANY,
|
||||
chan->language, args.format, args.timezone);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app_sayunixtime);
|
||||
res |= ast_unregister_application(app_datetime);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static 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;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Say time");
|
||||
@@ -1,143 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/translate.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/utils.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/manager.h"
|
||||
|
||||
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";
|
||||
|
||||
|
||||
static int senddtmf_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *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;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
digits = ast_strdupa(data);
|
||||
|
||||
if ((to = strchr(digits,'|'))) {
|
||||
*to = '\0';
|
||||
to++;
|
||||
timeout = atoi(to);
|
||||
}
|
||||
|
||||
if (timeout <= 0)
|
||||
timeout = 250;
|
||||
|
||||
res = ast_dtmf_stream(chan,NULL,digits,timeout);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static char mandescr_playdtmf[] =
|
||||
"Description: Plays a dtmf digit on the specified channel.\n"
|
||||
"Variables: (all are required)\n"
|
||||
" Channel: Channel name to send digit to\n"
|
||||
" Digit: The dtmf digit to play\n";
|
||||
|
||||
static int manager_play_dtmf(struct mansession *s, const struct message *m)
|
||||
{
|
||||
const char *channel = astman_get_header(m, "Channel");
|
||||
const char *digit = astman_get_header(m, "Digit");
|
||||
struct ast_channel *chan = ast_get_channel_by_name_locked(channel);
|
||||
|
||||
if (!chan) {
|
||||
astman_send_error(s, m, "Channel not specified");
|
||||
return 0;
|
||||
}
|
||||
if (!digit) {
|
||||
astman_send_error(s, m, "No digit specified");
|
||||
ast_mutex_unlock(&chan->lock);
|
||||
return 0;
|
||||
}
|
||||
|
||||
ast_senddigit(chan, *digit);
|
||||
|
||||
ast_mutex_unlock(&chan->lock);
|
||||
astman_send_ack(s, m, "DTMF successfully queued");
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
res |= ast_manager_unregister("PlayDTMF");
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_manager_register2( "PlayDTMF", EVENT_FLAG_CALL, manager_play_dtmf, "Play DTMF signal on a specific channel.", mandescr_playdtmf );
|
||||
res |= ast_register_application(app, senddtmf_exec, synopsis, descrip);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Send DTMF digits Application");
|
||||
@@ -1,130 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \note Requires support of sending text messages from channel driver
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/translate.h"
|
||||
#include "asterisk/image.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/app.h"
|
||||
|
||||
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";
|
||||
|
||||
|
||||
static int sendtext_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *u;
|
||||
char *status = "UNSUPPORTED";
|
||||
char *parse = NULL;
|
||||
int priority_jump = 0;
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(text);
|
||||
AST_APP_ARG(options);
|
||||
);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "SendText requires an argument (text[|options])\n");
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
} else
|
||||
parse = ast_strdupa(data);
|
||||
|
||||
AST_STANDARD_APP_ARGS(args, parse);
|
||||
|
||||
if (args.options) {
|
||||
if (strchr(args.options, 'j'))
|
||||
priority_jump = 1;
|
||||
}
|
||||
|
||||
ast_channel_lock(chan);
|
||||
if (!chan->tech->send_text) {
|
||||
ast_channel_unlock(chan);
|
||||
/* Does not support transport */
|
||||
if (priority_jump || ast_opt_priority_jumping)
|
||||
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
status = "FAILURE";
|
||||
ast_channel_unlock(chan);
|
||||
res = ast_sendtext(chan, args.text);
|
||||
if (!res)
|
||||
status = "SUCCESS";
|
||||
pbx_builtin_setvar_helper(chan, "SENDTEXTSTATUS", status);
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, sendtext_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Send Text Applications");
|
||||
@@ -1,159 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/translate.h"
|
||||
#include "asterisk/image.h"
|
||||
#include "asterisk/callerid.h"
|
||||
|
||||
static char *app2 = "SetCallerPres";
|
||||
|
||||
static char *synopsis2 = "Set CallerID Presentation";
|
||||
|
||||
|
||||
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 ast_module_user *u;
|
||||
int pres = -1;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
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);
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
|
||||
chan->cid.cid_pres = pres;
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
|
||||
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 ast_module_user *u;
|
||||
char *opt;
|
||||
int anitoo = 0;
|
||||
static int dep_warning = 0;
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "SetCallerID requires an argument!\n");
|
||||
return 0;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (!dep_warning) {
|
||||
dep_warning = 1;
|
||||
ast_log(LOG_WARNING, "SetCallerID is deprecated. Please use Set(CALLERID(all)=...) or Set(CALLERID(ani)=...) instead.\n");
|
||||
}
|
||||
|
||||
tmp = ast_strdupa(data);
|
||||
|
||||
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);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app2);
|
||||
res |= ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static 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;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Set CallerID Application");
|
||||
@@ -1,175 +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
|
||||
*
|
||||
* \author Justin Huff <jjhuff@mspin.net>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <sys/types.h>
|
||||
#include <stdlib.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/cdr.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/config.h"
|
||||
#include "asterisk/manager.h"
|
||||
#include "asterisk/utils.h"
|
||||
|
||||
|
||||
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"
|
||||
"\nThis application is deprecated in favor of Set(CDR(userfield)=...)\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"
|
||||
"\nThis application is deprecated in favor of Set(CDR(userfield)=...)\n";
|
||||
|
||||
static char *appendcdruserfield_app = "AppendCDRUserField";
|
||||
static char *appendcdruserfield_synopsis = "Append to the CDR user field";
|
||||
|
||||
|
||||
static int action_setcdruserfield(struct mansession *s, const struct message *m)
|
||||
{
|
||||
struct ast_channel *c = NULL;
|
||||
const char *userfield = astman_get_header(m, "UserField");
|
||||
const char *channel = astman_get_header(m, "Channel");
|
||||
const 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 ast_module_user *u;
|
||||
int res = 0;
|
||||
static int dep_warning = 0;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (chan->cdr && data) {
|
||||
ast_cdr_setuserfield(chan, (char*)data);
|
||||
}
|
||||
|
||||
if (!dep_warning) {
|
||||
dep_warning = 1;
|
||||
ast_log(LOG_WARNING, "SetCDRUserField is deprecated. Please use CDR(userfield) instead.\n");
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int appendcdruserfield_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *u;
|
||||
int res = 0;
|
||||
static int dep_warning = 0;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (chan->cdr && data) {
|
||||
ast_cdr_appenduserfield(chan, (char*)data);
|
||||
}
|
||||
|
||||
if (!dep_warning) {
|
||||
dep_warning = 1;
|
||||
ast_log(LOG_WARNING, "AppendCDRUserField is deprecated. Please use CDR(userfield) instead.\n");
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(setcdruserfield_app);
|
||||
res |= ast_unregister_application(appendcdruserfield_app);
|
||||
res |= ast_manager_unregister("SetCDRUserField");
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static 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;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "CDR user field apps");
|
||||
@@ -1,136 +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
|
||||
*
|
||||
* \author Frank Sautter - asterisk+at+sautter+dot+com
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <string.h>
|
||||
#include <stdlib.h>
|
||||
|
||||
#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";
|
||||
|
||||
|
||||
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"
|
||||
"This application is deprecated in favor of Set(CHANNEL(transfercapability)=...)\n"
|
||||
;
|
||||
|
||||
static int settransfercapability_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
char *tmp = NULL;
|
||||
struct ast_module_user *u;
|
||||
int x;
|
||||
char *opts;
|
||||
int transfercapability = -1;
|
||||
static int dep_warning = 0;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (!dep_warning) {
|
||||
dep_warning = 1;
|
||||
ast_log(LOG_WARNING, "SetTransferCapability is deprecated. Please use CHANNEL(transfercapability) instead.\n");
|
||||
}
|
||||
|
||||
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);
|
||||
ast_module_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);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, settransfercapability_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Set ISDN Transfer Capability");
|
||||
156
apps/app_skel.c
156
apps/app_skel.c
@@ -1,133 +1,75 @@
|
||||
/*
|
||||
* 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
|
||||
*
|
||||
* \author <Your Name Here> <<Your Email Here>>
|
||||
*
|
||||
* This is a skeleton for development of an Asterisk application
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
/*** MODULEINFO
|
||||
<defaultenabled>no</defaultenabled>
|
||||
***/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#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/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 *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";
|
||||
|
||||
enum {
|
||||
OPTION_A = (1 << 0),
|
||||
OPTION_B = (1 << 1),
|
||||
OPTION_C = (1 << 2),
|
||||
} option_flags;
|
||||
|
||||
enum {
|
||||
OPTION_ARG_B = 0,
|
||||
OPTION_ARG_C = 1,
|
||||
/* This *must* be the last value in this enum! */
|
||||
OPTION_ARG_ARRAY_SIZE = 2,
|
||||
} option_args;
|
||||
|
||||
AST_APP_OPTIONS(app_opts,{
|
||||
AST_APP_OPTION('a', OPTION_A),
|
||||
AST_APP_OPTION_ARG('b', OPTION_B, OPTION_ARG_B),
|
||||
AST_APP_OPTION_ARG('c', OPTION_C, OPTION_ARG_C),
|
||||
});
|
||||
#include <pthread.h>
|
||||
|
||||
|
||||
static int app_exec(struct ast_channel *chan, void *data)
|
||||
static char *tdesc = "Trivial skeleton Application";
|
||||
|
||||
static char *app = "skel";
|
||||
|
||||
STANDARD_LOCAL_USER;
|
||||
|
||||
LOCAL_USER_DECL;
|
||||
|
||||
static int skel_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_flags flags;
|
||||
struct ast_module_user *u;
|
||||
char *parse, *opts[OPTION_ARG_ARRAY_SIZE];
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(dummy);
|
||||
AST_APP_ARG(options);
|
||||
);
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "%s requires an argument (dummy|[options])\n", app);
|
||||
int res=0;
|
||||
struct localuser *u;
|
||||
if (!data) {
|
||||
ast_log(LOG_WARNING, "skel requires an argument (filename)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
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! */
|
||||
parse = ast_strdupa(data);
|
||||
|
||||
AST_STANDARD_APP_ARGS(args, parse);
|
||||
|
||||
if (args.argc == 2)
|
||||
ast_app_parse_options(app_opts, &flags, opts, args.options);
|
||||
|
||||
if (!ast_strlen_zero(args.dummy))
|
||||
ast_log(LOG_NOTICE, "Dummy value is : %s\n", args.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[OPTION_ARG_B] ? opts[OPTION_ARG_B] : "<unspecified>");
|
||||
|
||||
if (ast_test_flag(&flags, OPTION_C))
|
||||
ast_log(LOG_NOTICE, "Option C is set with : %s\n", opts[OPTION_ARG_C] ? opts[OPTION_ARG_C] : "<unspecified>");
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
int unload_module(void)
|
||||
{
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, skel_exec);
|
||||
}
|
||||
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
res = ast_unregister_application(app);
|
||||
return res;
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
char *key()
|
||||
{
|
||||
return ast_register_application(app, app_exec, synopsis, descrip);
|
||||
return ASTERISK_GPL_KEY;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Skeleton (sample) Application");
|
||||
|
||||
1536
apps/app_sms.c
1536
apps/app_sms.c
File diff suppressed because it is too large
Load Diff
@@ -1,121 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
#include <sys/types.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/lock.h"
|
||||
|
||||
static char *synopsis = "Soft Hangup Application";
|
||||
|
||||
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";
|
||||
|
||||
|
||||
static int softhangup_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *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;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
cdata = ast_strdupa(data);
|
||||
match = strsep(&cdata, "|");
|
||||
options = strsep(&cdata, "|");
|
||||
all = options && strchr(options,'a');
|
||||
c = ast_channel_walk_locked(NULL);
|
||||
while (c) {
|
||||
ast_copy_string(name, c->name, sizeof(name));
|
||||
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->tech->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);
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, softhangup_exec, synopsis, desc);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Hangs up the requested channel");
|
||||
@@ -1,858 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 2006, Digium, Inc.
|
||||
*
|
||||
* 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 Speech Recognition Utility Applications
|
||||
*
|
||||
* \author Joshua Colp <jcolp@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$");
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/speech.h"
|
||||
|
||||
/* Descriptions for each application */
|
||||
static char *speechcreate_descrip =
|
||||
"SpeechCreate(engine name)\n"
|
||||
"This application creates information to be used by all the other applications. It must be called before doing any speech recognition activities such as activating a grammar.\n"
|
||||
"It takes the engine name to use as the argument, if not specified the default engine will be used.\n";
|
||||
|
||||
static char *speechactivategrammar_descrip =
|
||||
"SpeechActivateGrammar(Grammar Name)\n"
|
||||
"This activates the specified grammar to be recognized by the engine. A grammar tells the speech recognition engine what to recognize, \n"
|
||||
"and how to portray it back to you in the dialplan. The grammar name is the only argument to this application.\n";
|
||||
|
||||
static char *speechstart_descrip =
|
||||
"SpeechStart()\n"
|
||||
"Tell the speech recognition engine that it should start trying to get results from audio being fed to it. This has no arguments.\n";
|
||||
|
||||
static char *speechbackground_descrip =
|
||||
"SpeechBackground(Sound File|Timeout)\n"
|
||||
"This application plays a sound file and waits for the person to speak. Once they start speaking playback of the file stops, and silence is heard.\n"
|
||||
"Once they stop talking the processing sound is played to indicate the speech recognition engine is working.\n"
|
||||
"Once results are available the application returns and results (score and text) are available using dialplan functions.\n"
|
||||
"The first text and score are ${SPEECH_TEXT(0)} AND ${SPEECH_SCORE(0)} while the second are ${SPEECH_TEXT(1)} and ${SPEECH_SCORE(1)}.\n"
|
||||
"The first argument is the sound file and the second is the timeout. Note the timeout will only start once the sound file has stopped playing.\n";
|
||||
|
||||
static char *speechdeactivategrammar_descrip =
|
||||
"SpeechDeactivateGrammar(Grammar Name)\n"
|
||||
"This deactivates the specified grammar so that it is no longer recognized. The only argument is the grammar name to deactivate.\n";
|
||||
|
||||
static char *speechprocessingsound_descrip =
|
||||
"SpeechProcessingSound(Sound File)\n"
|
||||
"This changes the processing sound that SpeechBackground plays back when the speech recognition engine is processing and working to get results.\n"
|
||||
"It takes the sound file as the only argument.\n";
|
||||
|
||||
static char *speechdestroy_descrip =
|
||||
"SpeechDestroy()\n"
|
||||
"This destroys the information used by all the other speech recognition applications.\n"
|
||||
"If you call this application but end up wanting to recognize more speech, you must call SpeechCreate\n"
|
||||
"again before calling any other application. It takes no arguments.\n";
|
||||
|
||||
static char *speechload_descrip =
|
||||
"SpeechLoadGrammar(Grammar Name|Path)\n"
|
||||
"Load a grammar only on the channel, not globally.\n"
|
||||
"It takes the grammar name as first argument and path as second.\n";
|
||||
|
||||
static char *speechunload_descrip =
|
||||
"SpeechUnloadGrammar(Grammar Name)\n"
|
||||
"Unload a grammar. It takes the grammar name as the only argument.\n";
|
||||
|
||||
/*! \brief Helper function used by datastores to destroy the speech structure upon hangup */
|
||||
static void destroy_callback(void *data)
|
||||
{
|
||||
struct ast_speech *speech = (struct ast_speech*)data;
|
||||
|
||||
if (speech == NULL) {
|
||||
return;
|
||||
}
|
||||
|
||||
/* Deallocate now */
|
||||
ast_speech_destroy(speech);
|
||||
|
||||
return;
|
||||
}
|
||||
|
||||
/*! \brief Static structure for datastore information */
|
||||
static const struct ast_datastore_info speech_datastore = {
|
||||
.type = "speech",
|
||||
.destroy = destroy_callback
|
||||
};
|
||||
|
||||
/*! \brief Helper function used to find the speech structure attached to a channel */
|
||||
static struct ast_speech *find_speech(struct ast_channel *chan)
|
||||
{
|
||||
struct ast_speech *speech = NULL;
|
||||
struct ast_datastore *datastore = NULL;
|
||||
|
||||
datastore = ast_channel_datastore_find(chan, &speech_datastore, NULL);
|
||||
if (datastore == NULL) {
|
||||
return NULL;
|
||||
}
|
||||
speech = datastore->data;
|
||||
|
||||
return speech;
|
||||
}
|
||||
|
||||
/* Helper function to find a specific speech recognition result by number and nbest alternative */
|
||||
static struct ast_speech_result *find_result(struct ast_speech_result *results, char *result_num)
|
||||
{
|
||||
struct ast_speech_result *result = results;
|
||||
char *tmp = NULL;
|
||||
int nbest_num = 0, wanted_num = 0, i = 0;
|
||||
|
||||
if (!result)
|
||||
return NULL;
|
||||
|
||||
if ((tmp = strchr(result_num, '/'))) {
|
||||
*tmp++ = '\0';
|
||||
nbest_num = atoi(result_num);
|
||||
wanted_num = atoi(tmp);
|
||||
} else {
|
||||
wanted_num = atoi(result_num);
|
||||
}
|
||||
|
||||
do {
|
||||
if (result->nbest_num != nbest_num)
|
||||
continue;
|
||||
if (i == wanted_num)
|
||||
break;
|
||||
i++;
|
||||
} while ((result = result->next));
|
||||
|
||||
return result;
|
||||
}
|
||||
|
||||
/*! \brief SPEECH_SCORE() Dialplan Function */
|
||||
static int speech_score(struct ast_channel *chan, char *cmd, char *data,
|
||||
char *buf, size_t len)
|
||||
{
|
||||
struct ast_speech_result *result = NULL;
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
char tmp[128] = "";
|
||||
|
||||
if (data == NULL || speech == NULL || !(result = find_result(speech->results, data)))
|
||||
return -1;
|
||||
|
||||
snprintf(tmp, sizeof(tmp), "%d", result->score);
|
||||
|
||||
ast_copy_string(buf, tmp, len);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static struct ast_custom_function speech_score_function = {
|
||||
.name = "SPEECH_SCORE",
|
||||
.synopsis = "Gets the confidence score of a result.",
|
||||
.syntax = "SPEECH_SCORE([nbest number/]result number)",
|
||||
.desc =
|
||||
"Gets the confidence score of a result.\n",
|
||||
.read = speech_score,
|
||||
.write = NULL,
|
||||
};
|
||||
|
||||
/*! \brief SPEECH_TEXT() Dialplan Function */
|
||||
static int speech_text(struct ast_channel *chan, char *cmd, char *data,
|
||||
char *buf, size_t len)
|
||||
{
|
||||
struct ast_speech_result *result = NULL;
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
|
||||
if (data == NULL || speech == NULL || !(result = find_result(speech->results, data)))
|
||||
return -1;
|
||||
|
||||
if (result->text != NULL)
|
||||
ast_copy_string(buf, result->text, len);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static struct ast_custom_function speech_text_function = {
|
||||
.name = "SPEECH_TEXT",
|
||||
.synopsis = "Gets the recognized text of a result.",
|
||||
.syntax = "SPEECH_TEXT([nbest number/]result number)",
|
||||
.desc =
|
||||
"Gets the recognized text of a result.\n",
|
||||
.read = speech_text,
|
||||
.write = NULL,
|
||||
};
|
||||
|
||||
/*! \brief SPEECH_GRAMMAR() Dialplan Function */
|
||||
static int speech_grammar(struct ast_channel *chan, char *cmd, char *data,
|
||||
char *buf, size_t len)
|
||||
{
|
||||
struct ast_speech_result *result = NULL;
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
|
||||
if (data == NULL || speech == NULL || !(result = find_result(speech->results, data)))
|
||||
return -1;
|
||||
|
||||
if (result->grammar != NULL)
|
||||
ast_copy_string(buf, result->grammar, len);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static struct ast_custom_function speech_grammar_function = {
|
||||
.name = "SPEECH_GRAMMAR",
|
||||
.synopsis = "Gets the matched grammar of a result if available.",
|
||||
.syntax = "SPEECH_GRAMMAR([nbest number/]result number)",
|
||||
.desc =
|
||||
"Gets the matched grammar of a result if available.\n",
|
||||
.read = speech_grammar,
|
||||
.write = NULL,
|
||||
};
|
||||
|
||||
/*! \brief SPEECH_ENGINE() Dialplan Function */
|
||||
static int speech_engine_write(struct ast_channel *chan, char *cmd, char *data, const char *value)
|
||||
{
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
|
||||
if (data == NULL || speech == NULL)
|
||||
return -1;
|
||||
|
||||
ast_speech_change(speech, data, value);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static struct ast_custom_function speech_engine_function = {
|
||||
.name = "SPEECH_ENGINE",
|
||||
.synopsis = "Change a speech engine specific attribute.",
|
||||
.syntax = "SPEECH_ENGINE(name)=value",
|
||||
.desc =
|
||||
"Changes a speech engine specific attribute.\n",
|
||||
.read = NULL,
|
||||
.write = speech_engine_write,
|
||||
};
|
||||
|
||||
/*! \brief SPEECH_RESULTS_TYPE() Dialplan Function */
|
||||
static int speech_results_type_write(struct ast_channel *chan, char *cmd, char *data, const char *value)
|
||||
{
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
|
||||
if (data == NULL || speech == NULL)
|
||||
return -1;
|
||||
|
||||
if (!strcasecmp(value, "normal"))
|
||||
ast_speech_change_results_type(speech, AST_SPEECH_RESULTS_TYPE_NORMAL);
|
||||
else if (!strcasecmp(value, "nbest"))
|
||||
ast_speech_change_results_type(speech, AST_SPEECH_RESULTS_TYPE_NBEST);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static struct ast_custom_function speech_results_type_function = {
|
||||
.name = "SPEECH_RESULTS_TYPE",
|
||||
.synopsis = "Sets the type of results that will be returned.",
|
||||
.syntax = "SPEECH_RESULTS_TYPE()=results type",
|
||||
.desc =
|
||||
"Sets the type of results that will be returned. Valid options are normal or nbest.",
|
||||
.read = NULL,
|
||||
.write = speech_results_type_write,
|
||||
};
|
||||
|
||||
/*! \brief SPEECH() Dialplan Function */
|
||||
static int speech_read(struct ast_channel *chan, char *cmd, char *data,
|
||||
char *buf, size_t len)
|
||||
{
|
||||
int results = 0;
|
||||
struct ast_speech_result *result = NULL;
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
char tmp[128] = "";
|
||||
|
||||
/* Now go for the various options */
|
||||
if (!strcasecmp(data, "status")) {
|
||||
if (speech != NULL)
|
||||
ast_copy_string(buf, "1", len);
|
||||
else
|
||||
ast_copy_string(buf, "0", len);
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* Make sure we have a speech structure for everything else */
|
||||
if (speech == NULL) {
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* Check to see if they are checking for silence */
|
||||
if (!strcasecmp(data, "spoke")) {
|
||||
if (ast_test_flag(speech, AST_SPEECH_SPOKE))
|
||||
ast_copy_string(buf, "1", len);
|
||||
else
|
||||
ast_copy_string(buf, "0", len);
|
||||
} else if (!strcasecmp(data, "results")) {
|
||||
/* Count number of results */
|
||||
result = speech->results;
|
||||
while (result) {
|
||||
results++;
|
||||
result = result->next;
|
||||
}
|
||||
snprintf(tmp, sizeof(tmp), "%d", results);
|
||||
ast_copy_string(buf, tmp, len);
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static struct ast_custom_function speech_function = {
|
||||
.name = "SPEECH",
|
||||
.synopsis = "Gets information about speech recognition results.",
|
||||
.syntax = "SPEECH(argument)",
|
||||
.desc =
|
||||
"Gets information about speech recognition results.\n"
|
||||
"status: Returns 1 upon speech object existing, or 0 if not\n"
|
||||
"spoke: Returns 1 if spoker spoke, or 0 if not\n"
|
||||
"results: Returns number of results that were recognized\n",
|
||||
.read = speech_read,
|
||||
.write = NULL,
|
||||
};
|
||||
|
||||
|
||||
|
||||
/*! \brief SpeechCreate() Dialplan Application */
|
||||
static int speech_create(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *u = NULL;
|
||||
struct ast_speech *speech = NULL;
|
||||
struct ast_datastore *datastore = NULL;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
/* Request a speech object */
|
||||
speech = ast_speech_new(data, AST_FORMAT_SLINEAR);
|
||||
if (speech == NULL) {
|
||||
/* Not available */
|
||||
pbx_builtin_setvar_helper(chan, "ERROR", "1");
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
|
||||
datastore = ast_channel_datastore_alloc(&speech_datastore, NULL);
|
||||
if (datastore == NULL) {
|
||||
ast_speech_destroy(speech);
|
||||
pbx_builtin_setvar_helper(chan, "ERROR", "1");
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
datastore->data = speech;
|
||||
ast_channel_datastore_add(chan, datastore);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
/*! \brief SpeechLoadGrammar(Grammar Name|Path) Dialplan Application */
|
||||
static int speech_load(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0, argc = 0;
|
||||
struct ast_module_user *u = NULL;
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
char *argv[2], *args = NULL, *name = NULL, *path = NULL;
|
||||
|
||||
args = ast_strdupa(data);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (speech == NULL) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* Parse out arguments */
|
||||
argc = ast_app_separate_args(args, '|', argv, sizeof(argv) / sizeof(argv[0]));
|
||||
if (argc != 2) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
name = argv[0];
|
||||
path = argv[1];
|
||||
|
||||
/* Load the grammar locally on the object */
|
||||
res = ast_speech_grammar_load(speech, name, path);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
/*! \brief SpeechUnloadGrammar(Grammar Name) Dialplan Application */
|
||||
static int speech_unload(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *u = NULL;
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (speech == NULL) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* Unload the grammar */
|
||||
res = ast_speech_grammar_unload(speech, data);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
/*! \brief SpeechDeactivateGrammar(Grammar Name) Dialplan Application */
|
||||
static int speech_deactivate(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *u = NULL;
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (speech == NULL) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* Deactivate the grammar on the speech object */
|
||||
res = ast_speech_grammar_deactivate(speech, data);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
/*! \brief SpeechActivateGrammar(Grammar Name) Dialplan Application */
|
||||
static int speech_activate(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *u = NULL;
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (speech == NULL) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* Activate the grammar on the speech object */
|
||||
res = ast_speech_grammar_activate(speech, data);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
/*! \brief SpeechStart() Dialplan Application */
|
||||
static int speech_start(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *u = NULL;
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (speech == NULL) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
ast_speech_start(speech);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
/*! \brief SpeechProcessingSound(Sound File) Dialplan Application */
|
||||
static int speech_processing_sound(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *u = NULL;
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (speech == NULL) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
if (speech->processing_sound != NULL) {
|
||||
free(speech->processing_sound);
|
||||
speech->processing_sound = NULL;
|
||||
}
|
||||
|
||||
speech->processing_sound = strdup(data);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
/*! \brief Helper function used by speech_background to playback a soundfile */
|
||||
static int speech_streamfile(struct ast_channel *chan, const char *filename, const char *preflang)
|
||||
{
|
||||
struct ast_filestream *fs = NULL;
|
||||
|
||||
if (!(fs = ast_openstream(chan, filename, preflang)))
|
||||
return -1;
|
||||
|
||||
if (ast_applystream(chan, fs))
|
||||
return -1;
|
||||
|
||||
if (ast_playstream(fs))
|
||||
return -1;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
/*! \brief SpeechBackground(Sound File|Timeout) Dialplan Application */
|
||||
static int speech_background(struct ast_channel *chan, void *data)
|
||||
{
|
||||
unsigned int timeout = 0;
|
||||
int res = 0, done = 0, argc = 0, started = 0, quieted = 0, max_dtmf_len = 0;
|
||||
struct ast_module_user *u = NULL;
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
struct ast_frame *f = NULL;
|
||||
int oldreadformat = AST_FORMAT_SLINEAR;
|
||||
char dtmf[AST_MAX_EXTENSION] = "";
|
||||
time_t start, current;
|
||||
struct ast_datastore *datastore = NULL;
|
||||
char *argv[2], *args = NULL, *filename_tmp = NULL, *filename = NULL, tmp[2] = "";
|
||||
const char *tmp2 = NULL;
|
||||
|
||||
args = ast_strdupa(data);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (speech == NULL) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* If channel is not already answered, then answer it */
|
||||
if (chan->_state != AST_STATE_UP && ast_answer(chan)) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* Record old read format */
|
||||
oldreadformat = chan->readformat;
|
||||
|
||||
/* Change read format to be signed linear */
|
||||
if (ast_set_read_format(chan, AST_FORMAT_SLINEAR)) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* Parse out options */
|
||||
argc = ast_app_separate_args(args, '|', argv, sizeof(argv) / sizeof(argv[0]));
|
||||
if (argc > 0) {
|
||||
/* Yay sound file */
|
||||
filename_tmp = ast_strdupa(argv[0]);
|
||||
if (!ast_strlen_zero(argv[1])) {
|
||||
if ((timeout = atoi(argv[1])) == 0)
|
||||
timeout = -1;
|
||||
} else
|
||||
timeout = 0;
|
||||
}
|
||||
|
||||
/* See if the maximum DTMF length variable is set... we use a variable in case they want to carry it through their entire dialplan */
|
||||
if ((tmp2 = pbx_builtin_getvar_helper(chan, "SPEECH_DTMF_MAXLEN")) && !ast_strlen_zero(tmp2))
|
||||
max_dtmf_len = atoi(tmp2);
|
||||
|
||||
/* Before we go into waiting for stuff... make sure the structure is ready, if not - start it again */
|
||||
if (speech->state == AST_SPEECH_STATE_NOT_READY || speech->state == AST_SPEECH_STATE_DONE) {
|
||||
ast_speech_change_state(speech, AST_SPEECH_STATE_NOT_READY);
|
||||
ast_speech_start(speech);
|
||||
}
|
||||
|
||||
/* Ensure no streams are currently running */
|
||||
ast_stopstream(chan);
|
||||
|
||||
/* Okay it's streaming so go into a loop grabbing frames! */
|
||||
while (done == 0) {
|
||||
/* If the filename is null and stream is not running, start up a new sound file */
|
||||
if (!quieted && (chan->streamid == -1 && chan->timingfunc == NULL) && (filename = strsep(&filename_tmp, "&"))) {
|
||||
/* Discard old stream information */
|
||||
ast_stopstream(chan);
|
||||
/* Start new stream */
|
||||
speech_streamfile(chan, filename, chan->language);
|
||||
}
|
||||
|
||||
/* Run scheduled stuff */
|
||||
ast_sched_runq(chan->sched);
|
||||
|
||||
/* Yay scheduling */
|
||||
res = ast_sched_wait(chan->sched);
|
||||
if (res < 0) {
|
||||
res = 1000;
|
||||
}
|
||||
|
||||
/* If there is a frame waiting, get it - if not - oh well */
|
||||
if (ast_waitfor(chan, res) > 0) {
|
||||
f = ast_read(chan);
|
||||
if (f == NULL) {
|
||||
/* The channel has hung up most likely */
|
||||
done = 3;
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
/* Do timeout check (shared between audio/dtmf) */
|
||||
if ((!quieted || strlen(dtmf)) && started == 1) {
|
||||
time(¤t);
|
||||
if ((current-start) >= timeout) {
|
||||
done = 1;
|
||||
if (f)
|
||||
ast_frfree(f);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
/* Do checks on speech structure to see if it's changed */
|
||||
ast_mutex_lock(&speech->lock);
|
||||
if (ast_test_flag(speech, AST_SPEECH_QUIET)) {
|
||||
if (chan->stream)
|
||||
ast_stopstream(chan);
|
||||
ast_clear_flag(speech, AST_SPEECH_QUIET);
|
||||
quieted = 1;
|
||||
}
|
||||
/* Check state so we can see what to do */
|
||||
switch (speech->state) {
|
||||
case AST_SPEECH_STATE_READY:
|
||||
/* If audio playback has stopped do a check for timeout purposes */
|
||||
if (chan->streamid == -1 && chan->timingfunc == NULL)
|
||||
ast_stopstream(chan);
|
||||
if (!quieted && chan->stream == NULL && timeout && started == 0 && !filename_tmp) {
|
||||
if (timeout == -1) {
|
||||
done = 1;
|
||||
if (f)
|
||||
ast_frfree(f);
|
||||
break;
|
||||
}
|
||||
time(&start);
|
||||
started = 1;
|
||||
}
|
||||
/* Write audio frame out to speech engine if no DTMF has been received */
|
||||
if (!strlen(dtmf) && f != NULL && f->frametype == AST_FRAME_VOICE) {
|
||||
ast_speech_write(speech, f->data, f->datalen);
|
||||
}
|
||||
break;
|
||||
case AST_SPEECH_STATE_WAIT:
|
||||
/* Cue up waiting sound if not already playing */
|
||||
if (!strlen(dtmf)) {
|
||||
if (chan->stream == NULL) {
|
||||
if (speech->processing_sound != NULL) {
|
||||
if (strlen(speech->processing_sound) > 0 && strcasecmp(speech->processing_sound,"none")) {
|
||||
speech_streamfile(chan, speech->processing_sound, chan->language);
|
||||
}
|
||||
}
|
||||
} else if (chan->streamid == -1 && chan->timingfunc == NULL) {
|
||||
ast_stopstream(chan);
|
||||
if (speech->processing_sound != NULL) {
|
||||
if (strlen(speech->processing_sound) > 0 && strcasecmp(speech->processing_sound,"none")) {
|
||||
speech_streamfile(chan, speech->processing_sound, chan->language);
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
break;
|
||||
case AST_SPEECH_STATE_DONE:
|
||||
/* Now that we are done... let's switch back to not ready state */
|
||||
ast_speech_change_state(speech, AST_SPEECH_STATE_NOT_READY);
|
||||
if (!strlen(dtmf)) {
|
||||
/* Copy to speech structure the results, if available */
|
||||
speech->results = ast_speech_results_get(speech);
|
||||
/* Break out of our background too */
|
||||
done = 1;
|
||||
/* Stop audio playback */
|
||||
if (chan->stream != NULL) {
|
||||
ast_stopstream(chan);
|
||||
}
|
||||
}
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
ast_mutex_unlock(&speech->lock);
|
||||
|
||||
/* Deal with other frame types */
|
||||
if (f != NULL) {
|
||||
/* Free the frame we received */
|
||||
switch (f->frametype) {
|
||||
case AST_FRAME_DTMF:
|
||||
if (f->subclass == '#') {
|
||||
done = 1;
|
||||
} else {
|
||||
if (chan->stream != NULL) {
|
||||
ast_stopstream(chan);
|
||||
}
|
||||
if (!started) {
|
||||
/* Change timeout to be 5 seconds for DTMF input */
|
||||
timeout = (chan->pbx && chan->pbx->dtimeout) ? chan->pbx->dtimeout : 5;
|
||||
started = 1;
|
||||
}
|
||||
time(&start);
|
||||
snprintf(tmp, sizeof(tmp), "%c", f->subclass);
|
||||
strncat(dtmf, tmp, sizeof(dtmf));
|
||||
/* If the maximum length of the DTMF has been reached, stop now */
|
||||
if (max_dtmf_len && strlen(dtmf) == max_dtmf_len)
|
||||
done = 1;
|
||||
}
|
||||
break;
|
||||
case AST_FRAME_CONTROL:
|
||||
switch (f->subclass) {
|
||||
case AST_CONTROL_HANGUP:
|
||||
/* Since they hung up we should destroy the speech structure */
|
||||
done = 3;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
default:
|
||||
break;
|
||||
}
|
||||
ast_frfree(f);
|
||||
f = NULL;
|
||||
}
|
||||
}
|
||||
|
||||
if (strlen(dtmf)) {
|
||||
/* We sort of make a results entry */
|
||||
speech->results = ast_calloc(1, sizeof(*speech->results));
|
||||
if (speech->results != NULL) {
|
||||
speech->results->score = 1000;
|
||||
speech->results->text = strdup(dtmf);
|
||||
speech->results->grammar = strdup("dtmf");
|
||||
}
|
||||
}
|
||||
|
||||
/* See if it was because they hung up */
|
||||
if (done == 3) {
|
||||
/* Destroy speech structure */
|
||||
ast_speech_destroy(speech);
|
||||
datastore = ast_channel_datastore_find(chan, &speech_datastore, NULL);
|
||||
if (datastore != NULL) {
|
||||
ast_channel_datastore_remove(chan, datastore);
|
||||
}
|
||||
} else {
|
||||
/* Channel is okay so restore read format */
|
||||
ast_set_read_format(chan, oldreadformat);
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
||||
/*! \brief SpeechDestroy() Dialplan Application */
|
||||
static int speech_destroy(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *u = NULL;
|
||||
struct ast_speech *speech = find_speech(chan);
|
||||
struct ast_datastore *datastore = NULL;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (speech == NULL) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* Destroy speech structure */
|
||||
ast_speech_destroy(speech);
|
||||
|
||||
datastore = ast_channel_datastore_find(chan, &speech_datastore, NULL);
|
||||
if (datastore != NULL) {
|
||||
ast_channel_datastore_remove(chan, datastore);
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res = 0;
|
||||
|
||||
res = ast_unregister_application("SpeechCreate");
|
||||
res |= ast_unregister_application("SpeechLoadGrammar");
|
||||
res |= ast_unregister_application("SpeechUnloadGrammar");
|
||||
res |= ast_unregister_application("SpeechActivateGrammar");
|
||||
res |= ast_unregister_application("SpeechDeactivateGrammar");
|
||||
res |= ast_unregister_application("SpeechStart");
|
||||
res |= ast_unregister_application("SpeechBackground");
|
||||
res |= ast_unregister_application("SpeechDestroy");
|
||||
res |= ast_unregister_application("SpeechProcessingSound");
|
||||
res |= ast_custom_function_unregister(&speech_function);
|
||||
res |= ast_custom_function_unregister(&speech_score_function);
|
||||
res |= ast_custom_function_unregister(&speech_text_function);
|
||||
res |= ast_custom_function_unregister(&speech_grammar_function);
|
||||
res |= ast_custom_function_unregister(&speech_engine_function);
|
||||
res |= ast_custom_function_unregister(&speech_results_type_function);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
int res = 0;
|
||||
|
||||
res = ast_register_application("SpeechCreate", speech_create, "Create a Speech Structure", speechcreate_descrip);
|
||||
res |= ast_register_application("SpeechLoadGrammar", speech_load, "Load a Grammar", speechload_descrip);
|
||||
res |= ast_register_application("SpeechUnloadGrammar", speech_unload, "Unload a Grammar", speechunload_descrip);
|
||||
res |= ast_register_application("SpeechActivateGrammar", speech_activate, "Activate a Grammar", speechactivategrammar_descrip);
|
||||
res |= ast_register_application("SpeechDeactivateGrammar", speech_deactivate, "Deactivate a Grammar", speechdeactivategrammar_descrip);
|
||||
res |= ast_register_application("SpeechStart", speech_start, "Start recognizing voice in the audio stream", speechstart_descrip);
|
||||
res |= ast_register_application("SpeechBackground", speech_background, "Play a sound file and wait for speech to be recognized", speechbackground_descrip);
|
||||
res |= ast_register_application("SpeechDestroy", speech_destroy, "End speech recognition", speechdestroy_descrip);
|
||||
res |= ast_register_application("SpeechProcessingSound", speech_processing_sound, "Change background processing sound", speechprocessingsound_descrip);
|
||||
res |= ast_custom_function_register(&speech_function);
|
||||
res |= ast_custom_function_register(&speech_score_function);
|
||||
res |= ast_custom_function_register(&speech_text_function);
|
||||
res |= ast_custom_function_register(&speech_grammar_function);
|
||||
res |= ast_custom_function_register(&speech_engine_function);
|
||||
res |= ast_custom_function_register(&speech_results_type_function);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Dialplan Speech Applications");
|
||||
174
apps/app_stack.c
174
apps/app_stack.c
@@ -1,174 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (c) 2004-2006 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.
|
||||
*
|
||||
* \author Tilghman Lesher <app_stack_v002@the-tilghman.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#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 *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 = "Conditionally 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";
|
||||
|
||||
|
||||
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)
|
||||
{
|
||||
const 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 ast_module_user *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;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
snprintf(newlabel, sizeof(newlabel), "%s|%s|%d", chan->context, chan->exten, chan->priority + 1);
|
||||
|
||||
if (ast_parseable_goto(chan, data)) {
|
||||
ast_module_user_remove(u);
|
||||
return -1;
|
||||
}
|
||||
|
||||
pbx_builtin_pushvar_helper(chan, STACKVAR, newlabel);
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int gosubif_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *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(data);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
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);
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
ast_unregister_application(app_return);
|
||||
ast_unregister_application(app_pop);
|
||||
ast_unregister_application(app_gosubif);
|
||||
ast_unregister_application(app_gosub);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static 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;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Stack Routines");
|
||||
@@ -1,158 +1,96 @@
|
||||
/*
|
||||
* 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.
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief Execute arbitrary system commands
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#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 <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
#include <errno.h>
|
||||
#include <stdlib.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/options.h"
|
||||
#include <pthread.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";
|
||||
" System(command): Executes a command by using system(). Returns -1 on\n"
|
||||
"failure to execute the specified command. If the command itself executes\n"
|
||||
"but is in error, and if there exists a priority n + 101, where 'n' is the\n"
|
||||
"priority of the current instance, then the channel will be setup to\n"
|
||||
"continue at that priority level. Otherwise, System returns 0.\n";
|
||||
|
||||
static char *descrip2 =
|
||||
" TrySystem(command): Executes a command by using system().\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 ast_module_user *u;
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
struct localuser *u;
|
||||
if (!data) {
|
||||
ast_log(LOG_WARNING, "System requires an argument(command)\n");
|
||||
pbx_builtin_setvar_helper(chan, chanvar, "FAILURE");
|
||||
return failmode;
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
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 (ast_opt_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->callerid))
|
||||
chan->priority+=100;
|
||||
res = 0;
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
}
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int system_exec(struct ast_channel *chan, void *data)
|
||||
int unload_module(void)
|
||||
{
|
||||
return system_exec_helper(chan, data, -1);
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
static int trysystem_exec(struct ast_channel *chan, void *data)
|
||||
int load_module(void)
|
||||
{
|
||||
return system_exec_helper(chan, data, 0);
|
||||
return ast_register_application(app, skel_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
res |= ast_unregister_application(app2);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
char *key()
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_register_application(app2, trysystem_exec, synopsis2, descrip2);
|
||||
res |= ast_register_application(app, system_exec, synopsis, descrip);
|
||||
|
||||
return res;
|
||||
return ASTERISK_GPL_KEY;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Generic System() application");
|
||||
|
||||
@@ -1,227 +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
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/translate.h"
|
||||
#include "asterisk/utils.h"
|
||||
#include "asterisk/dsp.h"
|
||||
|
||||
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";
|
||||
|
||||
|
||||
static int background_detect_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *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;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
tmp = ast_strdupa(data);
|
||||
|
||||
stringp=tmp;
|
||||
strsep(&stringp, "|");
|
||||
options = strsep(&stringp, "|");
|
||||
if (options) {
|
||||
if ((sscanf(options, "%d", &x) == 1) && (x > 0))
|
||||
sil = x;
|
||||
options = strsep(&stringp, "|");
|
||||
if (options) {
|
||||
if ((sscanf(options, "%d", &x) == 1) && (x > 0))
|
||||
min = x;
|
||||
options = strsep(&stringp, "|");
|
||||
if (options) {
|
||||
if ((sscanf(options, "%d", &x) == 1) && (x > 0))
|
||||
max = x;
|
||||
}
|
||||
}
|
||||
}
|
||||
ast_log(LOG_DEBUG, "Preparing detect of '%s', sil=%d,min=%d,max=%d\n",
|
||||
tmp, sil, min, max);
|
||||
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);
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, background_detect_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Playback with Talk Detection");
|
||||
512
apps/app_test.c
512
apps/app_test.c
@@ -1,512 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 1999 - 2005, Digium, Inc.
|
||||
*
|
||||
* Mark Spencer <markster@digium.com>
|
||||
* Russell Bryant <russelb@clemson.edu>
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief Applications to test connection and produce report in text file
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
* \author Russell Bryant <russelb@clemson.edu>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
#include <fcntl.h>
|
||||
#include <sys/types.h>
|
||||
#include <sys/stat.h>
|
||||
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/app.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/utils.h"
|
||||
|
||||
static char *tests_descrip =
|
||||
"TestServer(): Perform test server function and write call report.\n"
|
||||
"Results stored in /var/log/asterisk/testreports/<testid>-server.txt";
|
||||
static char *tests_app = "TestServer";
|
||||
static char *tests_synopsis = "Execute Interface Test Server";
|
||||
|
||||
static char *testc_descrip =
|
||||
"TestClient(testid): Executes test client with given testid.\n"
|
||||
"Results stored in /var/log/asterisk/testreports/<testid>-client.txt";
|
||||
|
||||
static char *testc_app = "TestClient";
|
||||
static char *testc_synopsis = "Execute Interface Test Client";
|
||||
|
||||
static int measurenoise(struct ast_channel *chan, int ms, char *who)
|
||||
{
|
||||
int res=0;
|
||||
int mssofar;
|
||||
int noise=0;
|
||||
int samples=0;
|
||||
int x;
|
||||
short *foo;
|
||||
struct timeval start;
|
||||
struct ast_frame *f;
|
||||
int rformat;
|
||||
rformat = chan->readformat;
|
||||
if (ast_set_read_format(chan, AST_FORMAT_SLINEAR)) {
|
||||
ast_log(LOG_NOTICE, "Unable to set to linear mode!\n");
|
||||
return -1;
|
||||
}
|
||||
start = ast_tvnow();
|
||||
for(;;) {
|
||||
mssofar = ast_tvdiff_ms(ast_tvnow(), start);
|
||||
if (mssofar > ms)
|
||||
break;
|
||||
res = ast_waitfor(chan, ms - mssofar);
|
||||
if (res < 1)
|
||||
break;
|
||||
f = ast_read(chan);
|
||||
if (!f) {
|
||||
res = -1;
|
||||
break;
|
||||
}
|
||||
if ((f->frametype == AST_FRAME_VOICE) && (f->subclass == AST_FORMAT_SLINEAR)) {
|
||||
foo = (short *)f->data;
|
||||
for (x=0;x<f->samples;x++) {
|
||||
noise += abs(foo[x]);
|
||||
samples++;
|
||||
}
|
||||
}
|
||||
ast_frfree(f);
|
||||
}
|
||||
|
||||
if (rformat) {
|
||||
if (ast_set_read_format(chan, rformat)) {
|
||||
ast_log(LOG_NOTICE, "Unable to restore original format!\n");
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
if (res < 0)
|
||||
return res;
|
||||
if (!samples) {
|
||||
ast_log(LOG_NOTICE, "No samples were received from the other side!\n");
|
||||
return -1;
|
||||
}
|
||||
ast_log(LOG_DEBUG, "%s: Noise: %d, samples: %d, avg: %d\n", who, noise, samples, noise / samples);
|
||||
return (noise / samples);
|
||||
}
|
||||
|
||||
static int sendnoise(struct ast_channel *chan, int ms)
|
||||
{
|
||||
int res;
|
||||
res = ast_tonepair_start(chan, 1537, 2195, ms, 8192);
|
||||
if (!res) {
|
||||
res = ast_waitfordigit(chan, ms);
|
||||
ast_tonepair_stop(chan);
|
||||
}
|
||||
return res;
|
||||
}
|
||||
|
||||
static int testclient_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *u;
|
||||
int res = 0;
|
||||
char *testid=data;
|
||||
char fn[80];
|
||||
char serverver[80];
|
||||
FILE *f;
|
||||
|
||||
/* Check for test id */
|
||||
if (ast_strlen_zero(testid)) {
|
||||
ast_log(LOG_WARNING, "TestClient requires an argument - the test id\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (chan->_state != AST_STATE_UP)
|
||||
res = ast_answer(chan);
|
||||
|
||||
/* Wait a few just to be sure things get started */
|
||||
res = ast_safe_sleep(chan, 3000);
|
||||
/* Transmit client version */
|
||||
if (!res)
|
||||
res = ast_dtmf_stream(chan, NULL, "8378*1#", 0);
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "Transmit client version\n");
|
||||
|
||||
/* Read server version */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "Read server version\n");
|
||||
if (!res)
|
||||
res = ast_app_getdata(chan, NULL, serverver, sizeof(serverver) - 1, 0);
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "server version: %s\n", serverver);
|
||||
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
|
||||
if (!res)
|
||||
res = ast_safe_sleep(chan, 1000);
|
||||
/* Send test id */
|
||||
if (!res)
|
||||
res = ast_dtmf_stream(chan, NULL, testid, 0);
|
||||
if (!res)
|
||||
res = ast_dtmf_stream(chan, NULL, "#", 0);
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "send test identifier: %s\n", testid);
|
||||
|
||||
if ((res >=0) && (!ast_strlen_zero(testid))) {
|
||||
/* Make the directory to hold the test results in case it's not there */
|
||||
snprintf(fn, sizeof(fn), "%s/testresults", ast_config_AST_LOG_DIR);
|
||||
mkdir(fn, 0777);
|
||||
snprintf(fn, sizeof(fn), "%s/testresults/%s-client.txt", ast_config_AST_LOG_DIR, testid);
|
||||
if ((f = fopen(fn, "w+"))) {
|
||||
setlinebuf(f);
|
||||
fprintf(f, "CLIENTCHAN: %s\n", chan->name);
|
||||
fprintf(f, "CLIENTTEST ID: %s\n", testid);
|
||||
fprintf(f, "ANSWER: PASS\n");
|
||||
res = 0;
|
||||
|
||||
if (!res) {
|
||||
/* Step 1: Wait for "1" */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestClient: 2. Wait DTMF 1\n");
|
||||
res = ast_waitfordigit(chan, 3000);
|
||||
fprintf(f, "WAIT DTMF 1: %s\n", (res != '1') ? "FAIL" : "PASS");
|
||||
if (res == '1')
|
||||
res = 0;
|
||||
else
|
||||
res = -1;
|
||||
}
|
||||
if (!res)
|
||||
res = ast_safe_sleep(chan, 1000);
|
||||
if (!res) {
|
||||
/* Step 2: Send "2" */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestClient: 2. Send DTMF 2\n");
|
||||
res = ast_dtmf_stream(chan, NULL, "2", 0);
|
||||
fprintf(f, "SEND DTMF 2: %s\n", (res < 0) ? "FAIL" : "PASS");
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
}
|
||||
if (!res) {
|
||||
/* Step 3: Wait one second */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestClient: 3. Wait one second\n");
|
||||
res = ast_safe_sleep(chan, 1000);
|
||||
fprintf(f, "WAIT 1 SEC: %s\n", (res < 0) ? "FAIL" : "PASS");
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
}
|
||||
if (!res) {
|
||||
/* Step 4: Measure noise */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestClient: 4. Measure noise\n");
|
||||
res = measurenoise(chan, 5000, "TestClient");
|
||||
fprintf(f, "MEASURENOISE: %s (%d)\n", (res < 0) ? "FAIL" : "PASS", res);
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
}
|
||||
if (!res) {
|
||||
/* Step 5: Wait for "4" */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestClient: 5. Wait DTMF 4\n");
|
||||
res = ast_waitfordigit(chan, 3000);
|
||||
fprintf(f, "WAIT DTMF 4: %s\n", (res != '4') ? "FAIL" : "PASS");
|
||||
if (res == '4')
|
||||
res = 0;
|
||||
else
|
||||
res = -1;
|
||||
}
|
||||
if (!res) {
|
||||
/* Step 6: Transmit tone noise */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestClient: 6. Transmit tone\n");
|
||||
res = sendnoise(chan, 6000);
|
||||
fprintf(f, "SENDTONE: %s\n", (res < 0) ? "FAIL" : "PASS");
|
||||
}
|
||||
if (!res || (res == '5')) {
|
||||
/* Step 7: Wait for "5" */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestClient: 7. Wait DTMF 5\n");
|
||||
if (!res)
|
||||
res = ast_waitfordigit(chan, 3000);
|
||||
fprintf(f, "WAIT DTMF 5: %s\n", (res != '5') ? "FAIL" : "PASS");
|
||||
if (res == '5')
|
||||
res = 0;
|
||||
else
|
||||
res = -1;
|
||||
}
|
||||
if (!res) {
|
||||
/* Step 8: Wait one second */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestClient: 8. Wait one second\n");
|
||||
res = ast_safe_sleep(chan, 1000);
|
||||
fprintf(f, "WAIT 1 SEC: %s\n", (res < 0) ? "FAIL" : "PASS");
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
}
|
||||
if (!res) {
|
||||
/* Step 9: Measure noise */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestClient: 6. Measure tone\n");
|
||||
res = measurenoise(chan, 4000, "TestClient");
|
||||
fprintf(f, "MEASURETONE: %s (%d)\n", (res < 0) ? "FAIL" : "PASS", res);
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
}
|
||||
if (!res) {
|
||||
/* Step 10: Send "7" */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestClient: 7. Send DTMF 7\n");
|
||||
res = ast_dtmf_stream(chan, NULL, "7", 0);
|
||||
fprintf(f, "SEND DTMF 7: %s\n", (res < 0) ? "FAIL" : "PASS");
|
||||
if (res > 0)
|
||||
res =0;
|
||||
}
|
||||
if (!res) {
|
||||
/* Step 11: Wait for "8" */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestClient: 11. Wait DTMF 8\n");
|
||||
res = ast_waitfordigit(chan, 3000);
|
||||
fprintf(f, "WAIT DTMF 8: %s\n", (res != '8') ? "FAIL" : "PASS");
|
||||
if (res == '8')
|
||||
res = 0;
|
||||
else
|
||||
res = -1;
|
||||
}
|
||||
if (option_debug && !res ) {
|
||||
/* Step 12: Hangup! */
|
||||
ast_log(LOG_DEBUG, "TestClient: 12. Hangup\n");
|
||||
}
|
||||
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "-- TEST COMPLETE--\n");
|
||||
fprintf(f, "-- END TEST--\n");
|
||||
fclose(f);
|
||||
res = -1;
|
||||
} else
|
||||
res = -1;
|
||||
} else {
|
||||
ast_log(LOG_NOTICE, "Did not read a test ID on '%s'\n", chan->name);
|
||||
res = -1;
|
||||
}
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int testserver_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *u;
|
||||
int res = 0;
|
||||
char testid[80]="";
|
||||
char fn[80];
|
||||
FILE *f;
|
||||
u = ast_module_user_add(chan);
|
||||
if (chan->_state != AST_STATE_UP)
|
||||
res = ast_answer(chan);
|
||||
/* Read version */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "Read client version\n");
|
||||
if (!res)
|
||||
res = ast_app_getdata(chan, NULL, testid, sizeof(testid) - 1, 0);
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
if (option_debug) {
|
||||
ast_log(LOG_DEBUG, "client version: %s\n", testid);
|
||||
ast_log(LOG_DEBUG, "Transmit server version\n");
|
||||
}
|
||||
res = ast_safe_sleep(chan, 1000);
|
||||
if (!res)
|
||||
res = ast_dtmf_stream(chan, NULL, "8378*1#", 0);
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
|
||||
if (!res)
|
||||
res = ast_app_getdata(chan, NULL, testid, sizeof(testid) - 1, 0);
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "read test identifier: %s\n", testid);
|
||||
/* Check for sneakyness */
|
||||
if (strchr(testid, '/'))
|
||||
res = -1;
|
||||
if ((res >=0) && (!ast_strlen_zero(testid))) {
|
||||
/* Got a Test ID! Whoo hoo! */
|
||||
/* Make the directory to hold the test results in case it's not there */
|
||||
snprintf(fn, sizeof(fn), "%s/testresults", ast_config_AST_LOG_DIR);
|
||||
mkdir(fn, 0777);
|
||||
snprintf(fn, sizeof(fn), "%s/testresults/%s-server.txt", ast_config_AST_LOG_DIR, testid);
|
||||
if ((f = fopen(fn, "w+"))) {
|
||||
setlinebuf(f);
|
||||
fprintf(f, "SERVERCHAN: %s\n", chan->name);
|
||||
fprintf(f, "SERVERTEST ID: %s\n", testid);
|
||||
fprintf(f, "ANSWER: PASS\n");
|
||||
ast_log(LOG_DEBUG, "Processing Test ID '%s'\n", testid);
|
||||
res = ast_safe_sleep(chan, 1000);
|
||||
if (!res) {
|
||||
/* Step 1: Send "1" */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestServer: 1. Send DTMF 1\n");
|
||||
res = ast_dtmf_stream(chan, NULL, "1", 0);
|
||||
fprintf(f, "SEND DTMF 1: %s\n", (res < 0) ? "FAIL" : "PASS");
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
}
|
||||
if (!res) {
|
||||
/* Step 2: Wait for "2" */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestServer: 2. Wait DTMF 2\n");
|
||||
res = ast_waitfordigit(chan, 3000);
|
||||
fprintf(f, "WAIT DTMF 2: %s\n", (res != '2') ? "FAIL" : "PASS");
|
||||
if (res == '2')
|
||||
res = 0;
|
||||
else
|
||||
res = -1;
|
||||
}
|
||||
if (!res) {
|
||||
/* Step 3: Measure noise */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestServer: 3. Measure noise\n");
|
||||
res = measurenoise(chan, 6000, "TestServer");
|
||||
fprintf(f, "MEASURENOISE: %s (%d)\n", (res < 0) ? "FAIL" : "PASS", res);
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
}
|
||||
if (!res) {
|
||||
/* Step 4: Send "4" */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestServer: 4. Send DTMF 4\n");
|
||||
res = ast_dtmf_stream(chan, NULL, "4", 0);
|
||||
fprintf(f, "SEND DTMF 4: %s\n", (res < 0) ? "FAIL" : "PASS");
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
}
|
||||
|
||||
if (!res) {
|
||||
/* Step 5: Wait one second */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestServer: 5. Wait one second\n");
|
||||
res = ast_safe_sleep(chan, 1000);
|
||||
fprintf(f, "WAIT 1 SEC: %s\n", (res < 0) ? "FAIL" : "PASS");
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
}
|
||||
|
||||
if (!res) {
|
||||
/* Step 6: Measure noise */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestServer: 6. Measure tone\n");
|
||||
res = measurenoise(chan, 4000, "TestServer");
|
||||
fprintf(f, "MEASURETONE: %s (%d)\n", (res < 0) ? "FAIL" : "PASS", res);
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
}
|
||||
|
||||
if (!res) {
|
||||
/* Step 7: Send "5" */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestServer: 7. Send DTMF 5\n");
|
||||
res = ast_dtmf_stream(chan, NULL, "5", 0);
|
||||
fprintf(f, "SEND DTMF 5: %s\n", (res < 0) ? "FAIL" : "PASS");
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
}
|
||||
|
||||
if (!res) {
|
||||
/* Step 8: Transmit tone noise */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestServer: 8. Transmit tone\n");
|
||||
res = sendnoise(chan, 6000);
|
||||
fprintf(f, "SENDTONE: %s\n", (res < 0) ? "FAIL" : "PASS");
|
||||
}
|
||||
|
||||
if (!res || (res == '7')) {
|
||||
/* Step 9: Wait for "7" */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestServer: 9. Wait DTMF 7\n");
|
||||
if (!res)
|
||||
res = ast_waitfordigit(chan, 3000);
|
||||
fprintf(f, "WAIT DTMF 7: %s\n", (res != '7') ? "FAIL" : "PASS");
|
||||
if (res == '7')
|
||||
res = 0;
|
||||
else
|
||||
res = -1;
|
||||
}
|
||||
if (!res)
|
||||
res = ast_safe_sleep(chan, 1000);
|
||||
if (!res) {
|
||||
/* Step 10: Send "8" */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestServer: 10. Send DTMF 8\n");
|
||||
res = ast_dtmf_stream(chan, NULL, "8", 0);
|
||||
fprintf(f, "SEND DTMF 8: %s\n", (res < 0) ? "FAIL" : "PASS");
|
||||
if (res > 0)
|
||||
res = 0;
|
||||
}
|
||||
if (!res) {
|
||||
/* Step 11: Wait for hangup to arrive! */
|
||||
if (option_debug)
|
||||
ast_log(LOG_DEBUG, "TestServer: 11. Waiting for hangup\n");
|
||||
res = ast_safe_sleep(chan, 10000);
|
||||
fprintf(f, "WAIT HANGUP: %s\n", (res < 0) ? "PASS" : "FAIL");
|
||||
}
|
||||
|
||||
ast_log(LOG_NOTICE, "-- TEST COMPLETE--\n");
|
||||
fprintf(f, "-- END TEST--\n");
|
||||
fclose(f);
|
||||
res = -1;
|
||||
} else
|
||||
res = -1;
|
||||
} else {
|
||||
ast_log(LOG_NOTICE, "Did not read a test ID on '%s'\n", chan->name);
|
||||
res = -1;
|
||||
}
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(testc_app);
|
||||
res |= ast_unregister_application(tests_app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_register_application(testc_app, testclient_exec, testc_synopsis, testc_descrip);
|
||||
res |= ast_register_application(tests_app, testserver_exec, tests_synopsis, tests_descrip);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Interface Test Application");
|
||||
@@ -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 Transfer a caller
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* Requires transfer support from channel driver
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/app.h"
|
||||
|
||||
|
||||
static const char *app = "Transfer";
|
||||
|
||||
static const char *synopsis = "Transfer caller to remote extension";
|
||||
|
||||
static const char *descrip =
|
||||
" Transfer([Tech/]dest[|options]): Requests the remote caller be transferred\n"
|
||||
"to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only\n"
|
||||
"an incoming call with the same channel technology will be transfered.\n"
|
||||
"Note that for SIP, if you transfer before call is setup, a 302 redirect\n"
|
||||
"SIP message will be returned to the caller.\n"
|
||||
"\nThe result of the application will be reported in the TRANSFERSTATUS\n"
|
||||
"channel variable:\n"
|
||||
" SUCCESS Transfer succeeded\n"
|
||||
" FAILURE Transfer failed\n"
|
||||
" UNSUPPORTED Transfer unsupported by channel driver\n"
|
||||
"The option string many contain the following character:\n"
|
||||
"'j' -- jump to n+101 priority if the channel transfer attempt\n"
|
||||
" fails\n";
|
||||
|
||||
static int transfer_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res;
|
||||
int len;
|
||||
struct ast_module_user *u;
|
||||
char *slash;
|
||||
char *tech = NULL;
|
||||
char *dest = NULL;
|
||||
char *status;
|
||||
char *parse;
|
||||
int priority_jump = 0;
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(dest);
|
||||
AST_APP_ARG(options);
|
||||
);
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (ast_strlen_zero((char *)data)) {
|
||||
ast_log(LOG_WARNING, "Transfer requires an argument ([Tech/]destination[|options])\n");
|
||||
ast_module_user_remove(u);
|
||||
pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "FAILURE");
|
||||
return 0;
|
||||
} else
|
||||
parse = ast_strdupa(data);
|
||||
|
||||
AST_STANDARD_APP_ARGS(args, parse);
|
||||
|
||||
if (args.options) {
|
||||
if (strchr(args.options, 'j'))
|
||||
priority_jump = 1;
|
||||
}
|
||||
|
||||
dest = args.dest;
|
||||
|
||||
if ((slash = strchr(dest, '/')) && (len = (slash - dest))) {
|
||||
tech = dest;
|
||||
dest = slash + 1;
|
||||
/* Allow execution only if the Tech/destination agrees with the type of the channel */
|
||||
if (strncasecmp(chan->tech->type, tech, len)) {
|
||||
pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "FAILURE");
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
|
||||
/* Check if the channel supports transfer before we try it */
|
||||
if (!chan->tech->transfer) {
|
||||
pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "UNSUPPORTED");
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
|
||||
res = ast_transfer(chan, dest);
|
||||
|
||||
if (res < 0) {
|
||||
status = "FAILURE";
|
||||
if (priority_jump || ast_opt_priority_jumping)
|
||||
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
|
||||
res = 0;
|
||||
} else {
|
||||
status = "SUCCESS";
|
||||
res = 0;
|
||||
}
|
||||
|
||||
pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", status);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, transfer_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Transfer");
|
||||
218
apps/app_url.c
218
apps/app_url.c
@@ -1,173 +1,137 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
* Asterisk -- A telephony toolkit for Linux.
|
||||
*
|
||||
* Copyright (C) 1999 - 2005, Digium, Inc.
|
||||
* App to transmit a URL
|
||||
*
|
||||
* Copyright (C) 1999, Mark Spencer
|
||||
*
|
||||
* Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
* Mark Spencer <markster@linux-support.net>
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
*
|
||||
* \brief App to transmit a URL
|
||||
*
|
||||
* \author Mark Spencer <markster@digium.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <asterisk/file.h>
|
||||
#include <asterisk/logger.h>
|
||||
#include <asterisk/channel.h>
|
||||
#include <asterisk/pbx.h>
|
||||
#include <asterisk/module.h>
|
||||
#include <asterisk/translate.h>
|
||||
#include <asterisk/image.h>
|
||||
#include <string.h>
|
||||
#include <stdlib.h>
|
||||
#include <pthread.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/translate.h"
|
||||
#include "asterisk/image.h"
|
||||
#include "asterisk/options.h"
|
||||
static char *tdesc = "Send URL Applications";
|
||||
|
||||
static char *app = "SendURL";
|
||||
|
||||
static char *synopsis = "Send a URL";
|
||||
|
||||
static char *descrip =
|
||||
" SendURL(URL[|option]): Requests client go to URL (IAX2) or sends the \n"
|
||||
"URL to the client (other channels).\n"
|
||||
"Result is returned in the SENDURLSTATUS channel variable:\n"
|
||||
" SUCCESS URL successfully sent to client\n"
|
||||
" FAILURE Failed to send URL\n"
|
||||
" NOLOAD Client failed to load URL (wait enabled)\n"
|
||||
" UNSUPPORTED Channel does not support URL transport\n"
|
||||
"\n"
|
||||
"If the option 'wait' is specified, execution will wait for an\n"
|
||||
"acknowledgement that the URL has been loaded before continuing\n"
|
||||
"\n"
|
||||
"If jumping is specified as an option (the 'j' flag), the client does not\n"
|
||||
"support Asterisk \"html\" transport, and there exists a step with priority\n"
|
||||
"n + 101, then execution will continue at that step.\n"
|
||||
"\n"
|
||||
"SendURL continues normally if the URL was sent correctly or if the channel\n"
|
||||
"does not support HTML transport. Otherwise, the channel is hung up.\n";
|
||||
" SendURL(URL[|option]): Requests client go to URL. If the client\n"
|
||||
"does not support html transport, and there exists a step with\n"
|
||||
"priority n + 101, then execution will continue at that step.\n"
|
||||
"Otherwise, execution will continue at the next priority level.\n"
|
||||
"SendURL only returns 0 if the URL was sent correctly or if\n"
|
||||
"the channel does not support HTML transport, and -1 otherwise.\n"
|
||||
"If the option 'wait' is specified, execution will wait for an\n"
|
||||
"acknowledgement that the URL has been loaded before continuing\n"
|
||||
"and will return -1 if the peer is unable to load the URL\n";
|
||||
|
||||
STANDARD_LOCAL_USER;
|
||||
|
||||
LOCAL_USER_DECL;
|
||||
|
||||
static int sendurl_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct ast_module_user *u;
|
||||
char *tmp;
|
||||
struct localuser *u;
|
||||
char tmp[256];
|
||||
char *options;
|
||||
int local_option_wait=0;
|
||||
int local_option_jump = 0;
|
||||
int option_wait=0;
|
||||
struct ast_frame *f;
|
||||
char *stringp=NULL;
|
||||
char *status = "FAILURE";
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
if (!data || !strlen((char *)data)) {
|
||||
ast_log(LOG_WARNING, "SendURL requires an argument (URL)\n");
|
||||
pbx_builtin_setvar_helper(chan, "SENDURLSTATUS", status);
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
tmp = ast_strdupa(data);
|
||||
|
||||
stringp=tmp;
|
||||
strsep(&stringp, "|");
|
||||
options = strsep(&stringp, "|");
|
||||
strncpy(tmp, (char *)data, sizeof(tmp)-1);
|
||||
strtok(tmp, "|");
|
||||
options = strtok(NULL, "|");
|
||||
if (options && !strcasecmp(options, "wait"))
|
||||
local_option_wait = 1;
|
||||
if (options && !strcasecmp(options, "j"))
|
||||
local_option_jump = 1;
|
||||
|
||||
option_wait = 1;
|
||||
LOCAL_USER_ADD(u);
|
||||
if (!ast_channel_supports_html(chan)) {
|
||||
/* Does not support transport */
|
||||
if (local_option_jump || ast_opt_priority_jumping)
|
||||
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
|
||||
pbx_builtin_setvar_helper(chan, "SENDURLSTATUS", "UNSUPPORTED");
|
||||
ast_module_user_remove(u);
|
||||
if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
|
||||
chan->priority += 100;
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return 0;
|
||||
}
|
||||
res = ast_channel_sendurl(chan, tmp);
|
||||
if (res == -1) {
|
||||
pbx_builtin_setvar_helper(chan, "SENDURLSTATUS", "FAILURE");
|
||||
ast_module_user_remove(u);
|
||||
return res;
|
||||
}
|
||||
status = "SUCCESS";
|
||||
if (local_option_wait) {
|
||||
for(;;) {
|
||||
/* Wait for an event */
|
||||
res = ast_waitfor(chan, -1);
|
||||
if (res < 0)
|
||||
break;
|
||||
f = ast_read(chan);
|
||||
if (!f) {
|
||||
res = -1;
|
||||
status = "FAILURE";
|
||||
break;
|
||||
}
|
||||
if (f->frametype == AST_FRAME_HTML) {
|
||||
switch(f->subclass) {
|
||||
case AST_HTML_LDCOMPLETE:
|
||||
res = 0;
|
||||
ast_frfree(f);
|
||||
status = "NOLOAD";
|
||||
goto out;
|
||||
if (res > -1) {
|
||||
if (option_wait) {
|
||||
for(;;) {
|
||||
/* Wait for an event */
|
||||
res = ast_waitfor(chan, -1);
|
||||
if (res < 0)
|
||||
break;
|
||||
case AST_HTML_NOSUPPORT:
|
||||
/* Does not support transport */
|
||||
status ="UNSUPPORTED";
|
||||
if (local_option_jump || ast_opt_priority_jumping)
|
||||
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
|
||||
res = 0;
|
||||
ast_frfree(f);
|
||||
goto out;
|
||||
f = ast_read(chan);
|
||||
if (!f) {
|
||||
res = -1;
|
||||
break;
|
||||
default:
|
||||
ast_log(LOG_WARNING, "Don't know what to do with HTML subclass %d\n", f->subclass);
|
||||
};
|
||||
}
|
||||
if (f->frametype == AST_FRAME_HTML) {
|
||||
switch(f->subclass) {
|
||||
case AST_HTML_LDCOMPLETE:
|
||||
res = 0;
|
||||
ast_frfree(f);
|
||||
goto out;
|
||||
break;
|
||||
case AST_HTML_NOSUPPORT:
|
||||
/* Does not support transport */
|
||||
if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
|
||||
chan->priority += 100;
|
||||
res = 0;
|
||||
goto out;
|
||||
break;
|
||||
default:
|
||||
ast_log(LOG_WARNING, "Don't know what to do with HTML subclass %d\n", f->subclass);
|
||||
};
|
||||
}
|
||||
ast_frfree(f);
|
||||
}
|
||||
ast_frfree(f);
|
||||
}
|
||||
}
|
||||
}
|
||||
out:
|
||||
pbx_builtin_setvar_helper(chan, "SENDURLSTATUS", status);
|
||||
ast_module_user_remove(u);
|
||||
LOCAL_USER_REMOVE(u);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, sendurl_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Send URL Applications");
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
|
||||
char *key()
|
||||
{
|
||||
return ASTERISK_GPL_KEY;
|
||||
}
|
||||
|
||||
@@ -1,109 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 1999 - 2005, Digium, Inc.
|
||||
*
|
||||
* 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 UserEvent application -- send manager event
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
#include <string.h>
|
||||
#include <unistd.h>
|
||||
|
||||
#include "asterisk/lock.h"
|
||||
#include "asterisk/file.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/manager.h"
|
||||
#include "asterisk/app.h"
|
||||
|
||||
static char *app = "UserEvent";
|
||||
|
||||
static char *synopsis = "Send an arbitrary event to the manager interface";
|
||||
|
||||
static char *descrip =
|
||||
" UserEvent(eventname[|body]): Sends an arbitrary event to the manager\n"
|
||||
"interface, with an optional body representing additional arguments. The\n"
|
||||
"body may be specified as a | delimeted list of headers. Each additional\n"
|
||||
"argument will be placed on a new line in the event. The format of the\n"
|
||||
"event will be:\n"
|
||||
" Event: UserEvent\n"
|
||||
" UserEvent: <specified event name>\n"
|
||||
" [body]\n"
|
||||
"If no body is specified, only Event and UserEvent headers will be present.\n";
|
||||
|
||||
|
||||
static int userevent_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
struct ast_module_user *u;
|
||||
char *parse, buf[2048] = "";
|
||||
int x, buflen = 0;
|
||||
AST_DECLARE_APP_ARGS(args,
|
||||
AST_APP_ARG(eventname);
|
||||
AST_APP_ARG(extra)[100];
|
||||
);
|
||||
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_log(LOG_WARNING, "UserEvent requires an argument (eventname|optional event body)\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
parse = ast_strdupa(data);
|
||||
|
||||
AST_STANDARD_APP_ARGS(args, parse);
|
||||
|
||||
for (x = 0; x < args.argc - 1; x++) {
|
||||
ast_copy_string(buf + buflen, args.extra[x], sizeof(buf) - buflen - 2);
|
||||
buflen += strlen(args.extra[x]);
|
||||
ast_copy_string(buf + buflen, "\r\n", 3);
|
||||
buflen += 2;
|
||||
}
|
||||
|
||||
manager_event(EVENT_FLAG_USER, "UserEvent", "UserEvent: %s\r\n%s", args.eventname, buf);
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
return ast_register_application(app, userevent_exec, synopsis, descrip);
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Custom User Event Application");
|
||||
@@ -1,168 +0,0 @@
|
||||
/*
|
||||
* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (c) 2004 - 2005 Tilghman Lesher. All rights reserved.
|
||||
*
|
||||
* Tilghman Lesher <app_verbose_v001@the-tilghman.com>
|
||||
*
|
||||
* This code is released by the author with no restrictions on usage.
|
||||
*
|
||||
* 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 Verbose logging application
|
||||
*
|
||||
* \author Tilghman Lesher <app_verbose_v001@the-tilghman.com>
|
||||
*
|
||||
* \ingroup applications
|
||||
*/
|
||||
|
||||
#include "asterisk.h"
|
||||
|
||||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <unistd.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "asterisk/options.h"
|
||||
#include "asterisk/logger.h"
|
||||
#include "asterisk/channel.h"
|
||||
#include "asterisk/pbx.h"
|
||||
#include "asterisk/module.h"
|
||||
|
||||
static char *app_verbose = "Verbose";
|
||||
static char *verbose_synopsis = "Send arbitrary text to verbose output";
|
||||
static char *verbose_descrip =
|
||||
"Verbose([<level>|]<message>)\n"
|
||||
" level must be an integer value. If not specified, defaults to 0.\n";
|
||||
|
||||
static char *app_log = "Log";
|
||||
static char *log_synopsis = "Send arbitrary text to a selected log level";
|
||||
static char *log_descrip =
|
||||
"Log(<level>|<message>)\n"
|
||||
" level must be one of ERROR, WARNING, NOTICE, DEBUG, VERBOSE, DTMF\n";
|
||||
|
||||
|
||||
static int verbose_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
char *vtext;
|
||||
int vsize;
|
||||
struct ast_module_user *u;
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
|
||||
if (data) {
|
||||
char *tmp;
|
||||
vtext = ast_strdupa(data);
|
||||
tmp = strsep(&vtext, "|");
|
||||
if (vtext) {
|
||||
if (sscanf(tmp, "%d", &vsize) != 1) {
|
||||
vsize = 0;
|
||||
ast_log(LOG_WARNING, "'%s' is not a verboser number\n", vtext);
|
||||
}
|
||||
} else {
|
||||
vtext = tmp;
|
||||
vsize = 0;
|
||||
}
|
||||
if (option_verbose >= vsize) {
|
||||
switch (vsize) {
|
||||
case 0:
|
||||
ast_verbose("%s\n", vtext);
|
||||
break;
|
||||
case 1:
|
||||
ast_verbose(VERBOSE_PREFIX_1 "%s\n", vtext);
|
||||
break;
|
||||
case 2:
|
||||
ast_verbose(VERBOSE_PREFIX_2 "%s\n", vtext);
|
||||
break;
|
||||
case 3:
|
||||
ast_verbose(VERBOSE_PREFIX_3 "%s\n", vtext);
|
||||
break;
|
||||
default:
|
||||
ast_verbose(VERBOSE_PREFIX_4 "%s\n", vtext);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
ast_module_user_remove(u);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int log_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
char *level, *ltext;
|
||||
struct ast_module_user *u;
|
||||
int lnum = -1;
|
||||
char extension[AST_MAX_EXTENSION + 5], context[AST_MAX_EXTENSION + 2];
|
||||
|
||||
u = ast_module_user_add(chan);
|
||||
if (ast_strlen_zero(data)) {
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
|
||||
ltext = ast_strdupa(data);
|
||||
|
||||
level = strsep(<ext, "|");
|
||||
|
||||
if (!strcasecmp(level, "ERROR")) {
|
||||
lnum = __LOG_ERROR;
|
||||
} else if (!strcasecmp(level, "WARNING")) {
|
||||
lnum = __LOG_WARNING;
|
||||
} else if (!strcasecmp(level, "NOTICE")) {
|
||||
lnum = __LOG_NOTICE;
|
||||
} else if (!strcasecmp(level, "DEBUG")) {
|
||||
lnum = __LOG_DEBUG;
|
||||
} else if (!strcasecmp(level, "VERBOSE")) {
|
||||
lnum = __LOG_VERBOSE;
|
||||
} else if (!strcasecmp(level, "DTMF")) {
|
||||
lnum = __LOG_DTMF;
|
||||
} else if (!strcasecmp(level, "EVENT")) {
|
||||
lnum = __LOG_EVENT;
|
||||
} else {
|
||||
ast_log(LOG_ERROR, "Unknown log level: '%s'\n", level);
|
||||
}
|
||||
|
||||
if (lnum > -1) {
|
||||
snprintf(context, sizeof(context), "@ %s", chan->context);
|
||||
snprintf(extension, sizeof(extension), "Ext. %s", chan->exten);
|
||||
|
||||
ast_log(lnum, extension, chan->priority, context, "%s\n", ltext);
|
||||
}
|
||||
ast_module_user_remove(u);
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_unregister_application(app_verbose);
|
||||
res |= ast_unregister_application(app_log);
|
||||
|
||||
ast_module_user_hangup_all();
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
int res;
|
||||
|
||||
res = ast_register_application(app_log, log_exec, log_synopsis, log_descrip);
|
||||
res |= ast_register_application(app_verbose, verbose_exec, verbose_synopsis, verbose_descrip);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Send verbose output");
|
||||
Some files were not shown because too many files have changed in this diff Show More
Reference in New Issue
Block a user