2004-08-27 22:53:56 +00:00
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/*
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2005-09-14 20:46:50 +00:00
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* Asterisk -- An open source telephony toolkit.
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1999-12-04 21:35:07 +00:00
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*
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2006-01-03 22:16:23 +00:00
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* Copyright (C) 1999 - 2006, Digium, Inc.
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1999-12-04 21:35:07 +00:00
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*
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2004-08-27 03:28:32 +00:00
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* Mark Spencer <markster@digium.com>
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1999-12-04 21:35:07 +00:00
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*
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2005-09-14 20:46:50 +00:00
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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1999-12-04 21:35:07 +00:00
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* This program is free software, distributed under the terms of
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2005-09-14 20:46:50 +00:00
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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2005-10-24 20:12:06 +00:00
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/*! \file
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2005-09-14 20:46:50 +00:00
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*
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2005-10-24 20:12:06 +00:00
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* \brief Channel Management
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2005-12-30 21:18:06 +00:00
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*
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* \author Mark Spencer <markster@digium.com>
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1999-12-04 21:35:07 +00:00
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*/
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2006-06-07 18:54:56 +00:00
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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2007-11-20 22:18:21 +00:00
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#include "asterisk/_private.h"
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1999-12-04 21:35:07 +00:00
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#include <sys/time.h>
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#include <signal.h>
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2006-06-07 18:54:56 +00:00
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#include <math.h>
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2006-04-24 17:11:45 +00:00
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2007-11-20 23:16:15 +00:00
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#include "asterisk/paths.h" /* use ast_config_AST_SYSTEM_NAME */
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2005-04-22 13:11:34 +00:00
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2005-04-21 06:02:45 +00:00
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#include "asterisk/pbx.h"
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#include "asterisk/frame.h"
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#include "asterisk/sched.h"
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#include "asterisk/channel.h"
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#include "asterisk/musiconhold.h"
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#include "asterisk/say.h"
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#include "asterisk/file.h"
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#include "asterisk/cli.h"
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#include "asterisk/translate.h"
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#include "asterisk/manager.h"
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#include "asterisk/chanvars.h"
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#include "asterisk/linkedlists.h"
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#include "asterisk/indications.h"
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#include "asterisk/monitor.h"
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#include "asterisk/causes.h"
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2005-07-25 22:13:32 +00:00
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#include "asterisk/callerid.h"
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2005-04-21 06:02:45 +00:00
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#include "asterisk/utils.h"
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#include "asterisk/lock.h"
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#include "asterisk/app.h"
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#include "asterisk/transcap.h"
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2005-07-08 21:14:34 +00:00
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#include "asterisk/devicestate.h"
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2006-02-03 22:37:29 +00:00
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#include "asterisk/sha1.h"
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Merge team/russell/ast_verbose_threadstorage
- instead of defining a free() wrapper in a bunch of files, define it as
ast_free() in utils.h and remove the copies from all the files.
- centralize and abstract the code used for doing thread storage. The code
lives in threadstorage.h, with one function being implemented in utils.c.
This new API includes generic thread storage as well as special functions
for handling thread local dynamic length string buffers.
- update ast_inet_ntoa() to use the new threadstorage API
- update ast_state2str() to use the new threadstorage API
- update ast_cli() to use the new threadstorage API
- Modify manager_event() to use thread storage. Instead of using a buffer of
4096 characters as the workspace for building the manager event, use a thread
local dynamic string. Now there is no length limitation on the length of the
body of a manager event.
- Significantly simplify the handling of ast_verbose() ...
- Instead of using a static char buffer and a lock to make sure only one
thread can be using ast_verbose() at a time, use a thread local dynamic
string as the workspace for preparing the verbose message. Instead of
locking around the entire function, the only locking done now is when the
message has been built and is being deliviered to the list of registered
verbose message handlers.
- This function was doing a strdup() on every message passed to it and
keeping a queue of the last 200 messages in memory. This has been
completely removed. The only place this was used was that if there were
any messages in the verbose queue when a verbose handler was registered,
all of the messages in the queue would be fed to it. So, I just made sure
that the console verbose handler and the network verbose handler (for
remote asterisk consoles) were registered before any verbose messages.
pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at
startup, but I didn't feel the performance hit of this message queue was
worth saving the initial verbose output for these very rarely used modules.
- I have removed the last three arguments to the verbose handlers, leaving
only the string itself because they aren't needed anymore. For example,
ast_verbose had some logic for telling the verbose handler to add
a newline if the buffer was completely full. Now that the buffer can grow
as needed, this doesn't matter anymore.
- remove unused function, ast_verbose_dmesg() which was to dispatch the
message queue
- Convert the list of verbose handlers to use the linked list macros.
- add missing newline characters to a few ast_verbose() calls
- convert the list of log channels to use the linked list macros in logger.c
- fix close_logger() to close all of the files it opened for logging
- update ast_log() to use a thread local dynamic string for its workspace
for preparing log messages instead of a buffer of size BUFSIZ (8kB on my
system) allocated on the stack. The dynamic string in this case is limited
to only growing to a maximum size of BUFSIZ.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39272 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-08 06:32:04 +00:00
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#include "asterisk/threadstorage.h"
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2006-07-28 18:59:59 +00:00
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#include "asterisk/slinfactory.h"
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2007-08-08 19:30:52 +00:00
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#include "asterisk/audiohook.h"
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2008-06-13 12:45:50 +00:00
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#include "asterisk/timing.h"
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2006-07-28 18:59:59 +00:00
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2007-08-08 21:44:58 +00:00
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#ifdef HAVE_EPOLL
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#include <sys/epoll.h>
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#endif
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struct ast_epoll_data {
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struct ast_channel *chan;
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int which;
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};
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2003-08-28 20:02:10 +00:00
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/* uncomment if you have problems with 'monitoring' synchronized files */
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#if 0
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#define MONITOR_CONSTANT_DELAY
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2007-02-24 20:29:41 +00:00
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#define MONITOR_DELAY 150 * 8 /*!< 150 ms of MONITORING DELAY */
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2003-08-28 20:02:10 +00:00
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#endif
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1999-12-04 21:35:07 +00:00
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2007-02-24 20:29:41 +00:00
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/*! \brief Prevent new channel allocation if shutting down. */
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2006-12-27 22:14:33 +00:00
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static int shutting_down;
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2005-06-06 02:29:18 +00:00
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2006-12-27 22:14:33 +00:00
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static int uniqueint;
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2000-01-02 20:59:00 +00:00
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2006-12-27 22:14:33 +00:00
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unsigned long global_fin, global_fout;
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2004-11-06 21:33:01 +00:00
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2006-10-19 01:00:57 +00:00
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AST_THREADSTORAGE(state2str_threadbuf);
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2006-07-21 22:36:25 +00:00
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#define STATE2STR_BUFSIZE 32
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2000-01-02 20:59:00 +00:00
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2007-04-24 19:03:16 +00:00
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/*! Default amount of time to use when emulating a digit as a begin and end
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* 100ms */
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#define AST_DEFAULT_EMULATE_DTMF_DURATION 100
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/*! Minimum allowed digit length - 80ms */
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#define AST_MIN_DTMF_DURATION 80
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/*! Minimum amount of time between the end of the last digit and the beginning
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* of a new one - 45ms */
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#define AST_MIN_DTMF_GAP 45
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2006-08-31 01:59:02 +00:00
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2007-02-24 20:29:41 +00:00
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/*! \brief List of channel drivers */
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1999-12-04 21:35:07 +00:00
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struct chanlist {
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2005-03-04 06:47:24 +00:00
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const struct ast_channel_tech *tech;
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2006-01-11 05:05:13 +00:00
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AST_LIST_ENTRY(chanlist) list;
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2005-06-06 02:29:18 +00:00
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};
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2008-02-18 04:43:33 +00:00
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#ifdef CHANNEL_TRACE
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/*! \brief Structure to hold channel context backtrace data */
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struct ast_chan_trace_data {
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int enabled;
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AST_LIST_HEAD_NOLOCK(, ast_chan_trace) trace;
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};
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/*! \brief Structure to save contexts where an ast_chan has been into */
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struct ast_chan_trace {
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char context[AST_MAX_CONTEXT];
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char exten[AST_MAX_EXTENSION];
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int priority;
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AST_LIST_ENTRY(ast_chan_trace) entry;
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};
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#endif
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2007-02-24 20:29:41 +00:00
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/*! \brief the list of registered channel types */
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2006-01-11 05:05:13 +00:00
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static AST_LIST_HEAD_NOLOCK_STATIC(backends, chanlist);
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2005-06-06 02:29:18 +00:00
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2007-02-24 20:29:41 +00:00
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/*! \brief the list of channels we have. Note that the lock for this list is used for
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2008-03-04 23:04:29 +00:00
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both the channels list and the backends list. */
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2007-06-11 14:41:39 +00:00
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static AST_RWLIST_HEAD_STATIC(channels, ast_channel);
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1999-12-04 21:35:07 +00:00
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2007-02-24 20:29:41 +00:00
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/*! \brief map AST_CAUSE's to readable string representations
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*
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* \ref causes.h
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*/
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2005-07-05 17:16:17 +00:00
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const struct ast_cause {
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int cause;
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2006-05-25 20:51:27 +00:00
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const char *name;
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2005-07-05 17:16:17 +00:00
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const char *desc;
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} causes[] = {
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2006-05-25 20:51:27 +00:00
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{ AST_CAUSE_UNALLOCATED, "UNALLOCATED", "Unallocated (unassigned) number" },
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{ AST_CAUSE_NO_ROUTE_TRANSIT_NET, "NO_ROUTE_TRANSIT_NET", "No route to specified transmit network" },
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{ AST_CAUSE_NO_ROUTE_DESTINATION, "NO_ROUTE_DESTINATION", "No route to destination" },
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{ AST_CAUSE_CHANNEL_UNACCEPTABLE, "CHANNEL_UNACCEPTABLE", "Channel unacceptable" },
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{ AST_CAUSE_CALL_AWARDED_DELIVERED, "CALL_AWARDED_DELIVERED", "Call awarded and being delivered in an established channel" },
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{ AST_CAUSE_NORMAL_CLEARING, "NORMAL_CLEARING", "Normal Clearing" },
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{ AST_CAUSE_USER_BUSY, "USER_BUSY", "User busy" },
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{ AST_CAUSE_NO_USER_RESPONSE, "NO_USER_RESPONSE", "No user responding" },
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{ AST_CAUSE_NO_ANSWER, "NO_ANSWER", "User alerting, no answer" },
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{ AST_CAUSE_CALL_REJECTED, "CALL_REJECTED", "Call Rejected" },
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{ AST_CAUSE_NUMBER_CHANGED, "NUMBER_CHANGED", "Number changed" },
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{ AST_CAUSE_DESTINATION_OUT_OF_ORDER, "DESTINATION_OUT_OF_ORDER", "Destination out of order" },
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{ AST_CAUSE_INVALID_NUMBER_FORMAT, "INVALID_NUMBER_FORMAT", "Invalid number format" },
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{ AST_CAUSE_FACILITY_REJECTED, "FACILITY_REJECTED", "Facility rejected" },
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{ AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY, "RESPONSE_TO_STATUS_ENQUIRY", "Response to STATus ENQuiry" },
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{ AST_CAUSE_NORMAL_UNSPECIFIED, "NORMAL_UNSPECIFIED", "Normal, unspecified" },
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{ AST_CAUSE_NORMAL_CIRCUIT_CONGESTION, "NORMAL_CIRCUIT_CONGESTION", "Circuit/channel congestion" },
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{ AST_CAUSE_NETWORK_OUT_OF_ORDER, "NETWORK_OUT_OF_ORDER", "Network out of order" },
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{ AST_CAUSE_NORMAL_TEMPORARY_FAILURE, "NORMAL_TEMPORARY_FAILURE", "Temporary failure" },
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{ AST_CAUSE_SWITCH_CONGESTION, "SWITCH_CONGESTION", "Switching equipment congestion" },
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{ AST_CAUSE_ACCESS_INFO_DISCARDED, "ACCESS_INFO_DISCARDED", "Access information discarded" },
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{ AST_CAUSE_REQUESTED_CHAN_UNAVAIL, "REQUESTED_CHAN_UNAVAIL", "Requested channel not available" },
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{ AST_CAUSE_PRE_EMPTED, "PRE_EMPTED", "Pre-empted" },
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{ AST_CAUSE_FACILITY_NOT_SUBSCRIBED, "FACILITY_NOT_SUBSCRIBED", "Facility not subscribed" },
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{ AST_CAUSE_OUTGOING_CALL_BARRED, "OUTGOING_CALL_BARRED", "Outgoing call barred" },
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{ AST_CAUSE_INCOMING_CALL_BARRED, "INCOMING_CALL_BARRED", "Incoming call barred" },
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{ AST_CAUSE_BEARERCAPABILITY_NOTAUTH, "BEARERCAPABILITY_NOTAUTH", "Bearer capability not authorized" },
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{ AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, "BEARERCAPABILITY_NOTAVAIL", "Bearer capability not available" },
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{ AST_CAUSE_BEARERCAPABILITY_NOTIMPL, "BEARERCAPABILITY_NOTIMPL", "Bearer capability not implemented" },
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{ AST_CAUSE_CHAN_NOT_IMPLEMENTED, "CHAN_NOT_IMPLEMENTED", "Channel not implemented" },
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{ AST_CAUSE_FACILITY_NOT_IMPLEMENTED, "FACILITY_NOT_IMPLEMENTED", "Facility not implemented" },
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{ AST_CAUSE_INVALID_CALL_REFERENCE, "INVALID_CALL_REFERENCE", "Invalid call reference value" },
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{ AST_CAUSE_INCOMPATIBLE_DESTINATION, "INCOMPATIBLE_DESTINATION", "Incompatible destination" },
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{ AST_CAUSE_INVALID_MSG_UNSPECIFIED, "INVALID_MSG_UNSPECIFIED", "Invalid message unspecified" },
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{ AST_CAUSE_MANDATORY_IE_MISSING, "MANDATORY_IE_MISSING", "Mandatory information element is missing" },
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{ AST_CAUSE_MESSAGE_TYPE_NONEXIST, "MESSAGE_TYPE_NONEXIST", "Message type nonexist." },
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{ AST_CAUSE_WRONG_MESSAGE, "WRONG_MESSAGE", "Wrong message" },
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{ AST_CAUSE_IE_NONEXIST, "IE_NONEXIST", "Info. element nonexist or not implemented" },
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{ AST_CAUSE_INVALID_IE_CONTENTS, "INVALID_IE_CONTENTS", "Invalid information element contents" },
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{ AST_CAUSE_WRONG_CALL_STATE, "WRONG_CALL_STATE", "Message not compatible with call state" },
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{ AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE, "RECOVERY_ON_TIMER_EXPIRE", "Recover on timer expiry" },
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{ AST_CAUSE_MANDATORY_IE_LENGTH_ERROR, "MANDATORY_IE_LENGTH_ERROR", "Mandatory IE length error" },
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{ AST_CAUSE_PROTOCOL_ERROR, "PROTOCOL_ERROR", "Protocol error, unspecified" },
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{ AST_CAUSE_INTERWORKING, "INTERWORKING", "Interworking, unspecified" },
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2005-07-05 17:16:17 +00:00
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};
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2006-02-11 07:06:47 +00:00
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struct ast_variable *ast_channeltype_list(void)
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{
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struct chanlist *cl;
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struct ast_variable *var=NULL, *prev = NULL;
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AST_LIST_TRAVERSE(&backends, cl, list) {
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if (prev) {
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2007-08-29 20:55:40 +00:00
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if ((prev->next = ast_variable_new(cl->tech->type, cl->tech->description, "")))
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2006-02-11 07:06:47 +00:00
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prev = prev->next;
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} else {
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2007-08-29 20:55:40 +00:00
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var = ast_variable_new(cl->tech->type, cl->tech->description, "");
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2006-02-11 07:06:47 +00:00
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prev = var;
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}
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}
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return var;
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}
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2007-02-24 20:29:41 +00:00
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/*! \brief Show channel types - CLI command */
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2007-10-11 19:03:06 +00:00
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static char *handle_cli_core_show_channeltypes(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
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2004-12-31 00:04:41 +00:00
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{
|
2006-01-04 09:22:45 +00:00
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#define FORMAT "%-10.10s %-40.40s %-12.12s %-12.12s %-12.12s\n"
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2005-12-24 05:18:40 +00:00
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struct chanlist *cl;
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2006-01-04 09:22:45 +00:00
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int count_chan = 0;
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2007-10-11 19:03:06 +00:00
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switch (cmd) {
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case CLI_INIT:
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e->command = "core show channeltypes";
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e->usage =
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"Usage: core show channeltypes\n"
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" Lists available channel types registered in your\n"
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" Asterisk server.\n";
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return NULL;
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case CLI_GENERATE:
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return NULL;
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}
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if (a->argc != 3)
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return CLI_SHOWUSAGE;
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ast_cli(a->fd, FORMAT, "Type", "Description", "Devicestate", "Indications", "Transfer");
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ast_cli(a->fd, FORMAT, "----------", "-----------", "-----------", "-----------", "--------");
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2008-01-24 03:30:56 +00:00
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2008-01-28 00:56:14 +00:00
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AST_RWLIST_RDLOCK(&channels);
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2008-01-24 03:30:56 +00:00
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2006-01-11 05:05:13 +00:00
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AST_LIST_TRAVERSE(&backends, cl, list) {
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2007-10-11 19:03:06 +00:00
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ast_cli(a->fd, FORMAT, cl->tech->type, cl->tech->description,
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2006-04-06 15:40:12 +00:00
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|
|
(cl->tech->devicestate) ? "yes" : "no",
|
2005-07-25 22:13:32 +00:00
|
|
|
(cl->tech->indicate) ? "yes" : "no",
|
|
|
|
(cl->tech->transfer) ? "yes" : "no");
|
2006-01-04 09:22:45 +00:00
|
|
|
count_chan++;
|
2004-12-31 00:04:41 +00:00
|
|
|
}
|
2008-01-24 03:30:56 +00:00
|
|
|
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
2008-01-24 03:30:56 +00:00
|
|
|
|
2007-10-11 19:03:06 +00:00
|
|
|
ast_cli(a->fd, "----------\n%d channel drivers registered.\n", count_chan);
|
2008-01-24 03:30:56 +00:00
|
|
|
|
2007-10-11 19:03:06 +00:00
|
|
|
return CLI_SUCCESS;
|
2004-12-31 00:04:41 +00:00
|
|
|
|
2005-06-02 21:37:09 +00:00
|
|
|
#undef FORMAT
|
2007-10-11 19:03:06 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
static char *complete_channeltypes(struct ast_cli_args *a)
|
|
|
|
{
|
|
|
|
struct chanlist *cl;
|
|
|
|
int which = 0;
|
|
|
|
int wordlen;
|
|
|
|
char *ret = NULL;
|
2005-06-02 21:37:09 +00:00
|
|
|
|
2007-10-11 19:03:06 +00:00
|
|
|
if (a->pos != 3)
|
|
|
|
return NULL;
|
|
|
|
|
|
|
|
wordlen = strlen(a->word);
|
|
|
|
|
|
|
|
AST_LIST_TRAVERSE(&backends, cl, list) {
|
|
|
|
if (!strncasecmp(a->word, cl->tech->type, wordlen) && ++which > a->n) {
|
|
|
|
ret = ast_strdup(cl->tech->type);
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
return ret;
|
2004-12-31 00:04:41 +00:00
|
|
|
}
|
|
|
|
|
2007-02-24 20:29:41 +00:00
|
|
|
/*! \brief Show details about a channel driver - CLI command */
|
2007-10-11 19:03:06 +00:00
|
|
|
static char *handle_cli_core_show_channeltype(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
2006-01-22 02:23:38 +00:00
|
|
|
{
|
|
|
|
struct chanlist *cl = NULL;
|
|
|
|
|
2007-10-11 19:03:06 +00:00
|
|
|
switch (cmd) {
|
|
|
|
case CLI_INIT:
|
|
|
|
e->command = "core show channeltype";
|
|
|
|
e->usage =
|
|
|
|
"Usage: core show channeltype <name>\n"
|
|
|
|
" Show details about the specified channel type, <name>.\n";
|
|
|
|
return NULL;
|
|
|
|
case CLI_GENERATE:
|
|
|
|
return complete_channeltypes(a);
|
|
|
|
}
|
|
|
|
|
|
|
|
if (a->argc != 4)
|
|
|
|
return CLI_SHOWUSAGE;
|
2006-01-22 02:23:38 +00:00
|
|
|
|
2008-01-28 00:56:14 +00:00
|
|
|
AST_RWLIST_RDLOCK(&channels);
|
2006-01-22 02:23:38 +00:00
|
|
|
|
|
|
|
AST_LIST_TRAVERSE(&backends, cl, list) {
|
2007-10-11 19:03:06 +00:00
|
|
|
if (!strncasecmp(cl->tech->type, a->argv[3], strlen(cl->tech->type)))
|
2006-01-22 02:23:38 +00:00
|
|
|
break;
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
if (!cl) {
|
2007-10-11 19:03:06 +00:00
|
|
|
ast_cli(a->fd, "\n%s is not a registered channel driver.\n", a->argv[3]);
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
2007-10-11 19:03:06 +00:00
|
|
|
return CLI_FAILURE;
|
2006-04-06 15:40:12 +00:00
|
|
|
}
|
2006-01-22 02:23:38 +00:00
|
|
|
|
2007-10-11 19:03:06 +00:00
|
|
|
ast_cli(a->fd,
|
2006-01-22 02:23:38 +00:00
|
|
|
"-- Info about channel driver: %s --\n"
|
|
|
|
" Device State: %s\n"
|
|
|
|
" Indication: %s\n"
|
|
|
|
" Transfer : %s\n"
|
|
|
|
" Capabilities: %d\n"
|
2006-08-31 01:59:02 +00:00
|
|
|
" Digit Begin: %s\n"
|
|
|
|
" Digit End: %s\n"
|
2006-01-22 02:23:38 +00:00
|
|
|
" Send HTML : %s\n"
|
|
|
|
" Image Support: %s\n"
|
|
|
|
" Text Support: %s\n",
|
|
|
|
cl->tech->type,
|
|
|
|
(cl->tech->devicestate) ? "yes" : "no",
|
|
|
|
(cl->tech->indicate) ? "yes" : "no",
|
|
|
|
(cl->tech->transfer) ? "yes" : "no",
|
|
|
|
(cl->tech->capabilities) ? cl->tech->capabilities : -1,
|
2006-08-31 01:59:02 +00:00
|
|
|
(cl->tech->send_digit_begin) ? "yes" : "no",
|
|
|
|
(cl->tech->send_digit_end) ? "yes" : "no",
|
2006-01-22 02:23:38 +00:00
|
|
|
(cl->tech->send_html) ? "yes" : "no",
|
|
|
|
(cl->tech->send_image) ? "yes" : "no",
|
|
|
|
(cl->tech->send_text) ? "yes" : "no"
|
|
|
|
|
|
|
|
);
|
|
|
|
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
2007-10-11 19:03:06 +00:00
|
|
|
return CLI_SUCCESS;
|
2006-01-22 02:23:38 +00:00
|
|
|
}
|
|
|
|
|
2006-09-18 19:54:18 +00:00
|
|
|
static struct ast_cli_entry cli_channel[] = {
|
2007-10-22 20:05:18 +00:00
|
|
|
AST_CLI_DEFINE(handle_cli_core_show_channeltypes, "List available channel types"),
|
|
|
|
AST_CLI_DEFINE(handle_cli_core_show_channeltype, "Give more details on that channel type")
|
2006-09-18 19:54:18 +00:00
|
|
|
};
|
2006-01-22 02:23:38 +00:00
|
|
|
|
2008-02-18 04:43:33 +00:00
|
|
|
#ifdef CHANNEL_TRACE
|
|
|
|
/*! \brief Destructor for the channel trace datastore */
|
|
|
|
static void ast_chan_trace_destroy_cb(void *data)
|
|
|
|
{
|
|
|
|
struct ast_chan_trace *trace;
|
|
|
|
struct ast_chan_trace_data *traced = data;
|
|
|
|
while ((trace = AST_LIST_REMOVE_HEAD(&traced->trace, entry))) {
|
|
|
|
ast_free(trace);
|
|
|
|
}
|
|
|
|
ast_free(traced);
|
|
|
|
}
|
|
|
|
|
|
|
|
/*! \brief Datastore to put the linked list of ast_chan_trace and trace status */
|
|
|
|
const struct ast_datastore_info ast_chan_trace_datastore_info = {
|
|
|
|
.type = "ChanTrace",
|
|
|
|
.destroy = ast_chan_trace_destroy_cb
|
|
|
|
};
|
|
|
|
|
|
|
|
/*! \brief Put the channel backtrace in a string */
|
|
|
|
int ast_channel_trace_serialize(struct ast_channel *chan, struct ast_str **buf)
|
|
|
|
{
|
|
|
|
int total = 0;
|
|
|
|
struct ast_chan_trace *trace;
|
|
|
|
struct ast_chan_trace_data *traced;
|
|
|
|
struct ast_datastore *store;
|
|
|
|
|
|
|
|
ast_channel_lock(chan);
|
|
|
|
store = ast_channel_datastore_find(chan, &ast_chan_trace_datastore_info, NULL);
|
|
|
|
if (!store) {
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
return total;
|
|
|
|
}
|
|
|
|
traced = store->data;
|
2008-10-31 09:31:10 +00:00
|
|
|
ast_str_reset(*buf);
|
2008-02-18 04:43:33 +00:00
|
|
|
AST_LIST_TRAVERSE(&traced->trace, trace, entry) {
|
|
|
|
if (ast_str_append(buf, 0, "[%d] => %s, %s, %d\n", total, trace->context, trace->exten, trace->priority) < 0) {
|
|
|
|
ast_log(LOG_ERROR, "Data Buffer Size Exceeded!\n");
|
|
|
|
total = -1;
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
total++;
|
|
|
|
}
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
return total;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* !\brief Whether or not context tracing is enabled */
|
|
|
|
int ast_channel_trace_is_enabled(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
struct ast_datastore *store = ast_channel_datastore_find(chan, &ast_chan_trace_datastore_info, NULL);
|
|
|
|
if (!store)
|
|
|
|
return 0;
|
|
|
|
return ((struct ast_chan_trace_data *)store->data)->enabled;
|
|
|
|
}
|
|
|
|
|
|
|
|
/*! \brief Update the context backtrace data if tracing is enabled */
|
|
|
|
static int ast_channel_trace_data_update(struct ast_channel *chan, struct ast_chan_trace_data *traced)
|
|
|
|
{
|
|
|
|
struct ast_chan_trace *trace;
|
|
|
|
if (!traced->enabled)
|
|
|
|
return 0;
|
|
|
|
/* If the last saved context does not match the current one
|
|
|
|
OR we have not saved any context so far, then save the current context */
|
|
|
|
if ((!AST_LIST_EMPTY(&traced->trace) && strcasecmp(AST_LIST_FIRST(&traced->trace)->context, chan->context)) ||
|
|
|
|
(AST_LIST_EMPTY(&traced->trace))) {
|
|
|
|
/* Just do some debug logging */
|
|
|
|
if (AST_LIST_EMPTY(&traced->trace))
|
|
|
|
ast_log(LOG_DEBUG, "Setting initial trace context to %s\n", chan->context);
|
|
|
|
else
|
|
|
|
ast_log(LOG_DEBUG, "Changing trace context from %s to %s\n", AST_LIST_FIRST(&traced->trace)->context, chan->context);
|
|
|
|
/* alloc or bail out */
|
|
|
|
trace = ast_malloc(sizeof(*trace));
|
|
|
|
if (!trace)
|
|
|
|
return -1;
|
|
|
|
/* save the current location and store it in the trace list */
|
|
|
|
ast_copy_string(trace->context, chan->context, sizeof(trace->context));
|
|
|
|
ast_copy_string(trace->exten, chan->exten, sizeof(trace->exten));
|
|
|
|
trace->priority = chan->priority;
|
|
|
|
AST_LIST_INSERT_HEAD(&traced->trace, trace, entry);
|
|
|
|
}
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
/*! \brief Update the context backtrace if tracing is enabled */
|
|
|
|
int ast_channel_trace_update(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
struct ast_datastore *store = ast_channel_datastore_find(chan, &ast_chan_trace_datastore_info, NULL);
|
|
|
|
if (!store)
|
|
|
|
return 0;
|
|
|
|
return ast_channel_trace_data_update(chan, store->data);
|
|
|
|
}
|
|
|
|
|
|
|
|
/*! \brief Enable context tracing in the channel */
|
|
|
|
int ast_channel_trace_enable(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
struct ast_datastore *store = ast_channel_datastore_find(chan, &ast_chan_trace_datastore_info, NULL);
|
|
|
|
struct ast_chan_trace_data *traced;
|
|
|
|
if (!store) {
|
2008-08-13 17:36:15 +00:00
|
|
|
store = ast_datastore_alloc(&ast_chan_trace_datastore_info, "ChanTrace");
|
2008-02-18 04:43:33 +00:00
|
|
|
if (!store)
|
|
|
|
return -1;
|
|
|
|
traced = ast_calloc(1, sizeof(*traced));
|
|
|
|
if (!traced) {
|
2008-08-13 17:36:15 +00:00
|
|
|
ast_datastore_free(store);
|
2008-02-18 04:43:33 +00:00
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
store->data = traced;
|
|
|
|
AST_LIST_HEAD_INIT_NOLOCK(&traced->trace);
|
|
|
|
ast_channel_datastore_add(chan, store);
|
|
|
|
}
|
|
|
|
((struct ast_chan_trace_data *)store->data)->enabled = 1;
|
|
|
|
ast_channel_trace_data_update(chan, store->data);
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
/*! \brief Disable context tracing in the channel */
|
|
|
|
int ast_channel_trace_disable(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
struct ast_datastore *store = ast_channel_datastore_find(chan, &ast_chan_trace_datastore_info, NULL);
|
|
|
|
if (!store)
|
|
|
|
return 0;
|
|
|
|
((struct ast_chan_trace_data *)store->data)->enabled = 0;
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
#endif /* CHANNEL_TRACE */
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Checks to see if a channel is needing hang up */
|
2001-12-20 15:21:47 +00:00
|
|
|
int ast_check_hangup(struct ast_channel *chan)
|
|
|
|
{
|
2006-04-06 15:40:12 +00:00
|
|
|
if (chan->_softhangup) /* yes if soft hangup flag set */
|
2005-05-15 04:48:30 +00:00
|
|
|
return 1;
|
2006-04-06 15:40:12 +00:00
|
|
|
if (!chan->tech_pvt) /* yes if no technology private data */
|
2005-05-15 04:48:30 +00:00
|
|
|
return 1;
|
2008-05-01 23:06:23 +00:00
|
|
|
if (ast_tvzero(chan->whentohangup)) /* no if no hangup scheduled */
|
2005-05-15 04:48:30 +00:00
|
|
|
return 0;
|
2008-05-01 23:06:23 +00:00
|
|
|
if (ast_tvdiff_ms(chan->whentohangup, ast_tvnow()) > 0) /* no if hangup time has not come yet. */
|
2005-05-15 04:48:30 +00:00
|
|
|
return 0;
|
2006-04-06 15:40:12 +00:00
|
|
|
chan->_softhangup |= AST_SOFTHANGUP_TIMEOUT; /* record event */
|
2001-12-20 15:21:47 +00:00
|
|
|
return 1;
|
|
|
|
}
|
|
|
|
|
2003-08-11 21:10:52 +00:00
|
|
|
static int ast_check_hangup_locked(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
int res;
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_lock(chan);
|
2003-08-11 21:10:52 +00:00
|
|
|
res = ast_check_hangup(chan);
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_unlock(chan);
|
2003-08-11 21:10:52 +00:00
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Initiate system shutdown */
|
2002-06-21 01:40:13 +00:00
|
|
|
void ast_begin_shutdown(int hangup)
|
|
|
|
{
|
|
|
|
struct ast_channel *c;
|
|
|
|
shutting_down = 1;
|
|
|
|
if (hangup) {
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_RDLOCK(&channels);
|
2008-05-19 20:45:25 +00:00
|
|
|
AST_RWLIST_TRAVERSE(&channels, c, chan_list) {
|
2002-09-10 20:48:20 +00:00
|
|
|
ast_softhangup(c, AST_SOFTHANGUP_SHUTDOWN);
|
2008-05-19 20:45:25 +00:00
|
|
|
}
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
2002-06-21 01:40:13 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief returns number of active/allocated channels */
|
2002-06-21 01:40:13 +00:00
|
|
|
int ast_active_channels(void)
|
|
|
|
{
|
|
|
|
struct ast_channel *c;
|
|
|
|
int cnt = 0;
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_RDLOCK(&channels);
|
|
|
|
AST_RWLIST_TRAVERSE(&channels, c, chan_list)
|
2002-06-21 01:40:13 +00:00
|
|
|
cnt++;
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
2002-06-21 01:40:13 +00:00
|
|
|
return cnt;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Cancel a shutdown in progress */
|
2002-06-21 01:40:13 +00:00
|
|
|
void ast_cancel_shutdown(void)
|
|
|
|
{
|
|
|
|
shutting_down = 0;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Returns non-zero if Asterisk is being shut down */
|
2002-06-21 01:40:13 +00:00
|
|
|
int ast_shutting_down(void)
|
|
|
|
{
|
|
|
|
return shutting_down;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Set when to hangup channel */
|
2008-05-01 23:06:23 +00:00
|
|
|
void ast_channel_setwhentohangup_tv(struct ast_channel *chan, struct timeval offset)
|
2001-12-20 15:21:47 +00:00
|
|
|
{
|
2008-05-01 23:06:23 +00:00
|
|
|
chan->whentohangup = ast_tvzero(offset) ? offset : ast_tvadd(offset, ast_tvnow());
|
2006-04-06 15:40:12 +00:00
|
|
|
ast_queue_frame(chan, &ast_null_frame);
|
2001-12-20 15:21:47 +00:00
|
|
|
return;
|
|
|
|
}
|
|
|
|
|
2008-05-01 23:06:23 +00:00
|
|
|
void ast_channel_setwhentohangup(struct ast_channel *chan, time_t offset)
|
|
|
|
{
|
2008-08-10 19:35:50 +00:00
|
|
|
struct timeval when = { offset, };
|
|
|
|
ast_channel_setwhentohangup_tv(chan, when);
|
2008-05-01 23:06:23 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Compare a offset with when to hangup channel */
|
2008-05-01 23:06:23 +00:00
|
|
|
int ast_channel_cmpwhentohangup_tv(struct ast_channel *chan, struct timeval offset)
|
2005-10-04 23:28:57 +00:00
|
|
|
{
|
2008-05-01 23:06:23 +00:00
|
|
|
struct timeval whentohangup;
|
2005-10-04 23:28:57 +00:00
|
|
|
|
2008-05-01 23:06:23 +00:00
|
|
|
if (ast_tvzero(chan->whentohangup))
|
|
|
|
return ast_tvzero(offset) ? 0 : -1;
|
2007-06-12 13:58:13 +00:00
|
|
|
|
2008-05-01 23:06:23 +00:00
|
|
|
if (ast_tvzero(offset))
|
2007-06-12 13:58:13 +00:00
|
|
|
return 1;
|
|
|
|
|
2008-05-01 23:06:23 +00:00
|
|
|
whentohangup = ast_tvadd(offset, ast_tvnow());
|
2007-06-12 13:58:13 +00:00
|
|
|
|
2008-05-01 23:06:23 +00:00
|
|
|
return ast_tvdiff_ms(whentohangup, chan->whentohangup);
|
|
|
|
}
|
|
|
|
|
|
|
|
int ast_channel_cmpwhentohangup(struct ast_channel *chan, time_t offset)
|
|
|
|
{
|
2008-08-10 20:23:50 +00:00
|
|
|
struct timeval when = { offset, };
|
|
|
|
return ast_channel_cmpwhentohangup_tv(chan, when);
|
2005-10-04 23:28:57 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Register a new telephony channel in Asterisk */
|
2005-03-04 06:47:24 +00:00
|
|
|
int ast_channel_register(const struct ast_channel_tech *tech)
|
2003-03-30 22:55:42 +00:00
|
|
|
{
|
2005-03-04 06:47:24 +00:00
|
|
|
struct chanlist *chan;
|
|
|
|
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_WRLOCK(&channels);
|
2003-03-30 22:55:42 +00:00
|
|
|
|
2006-01-11 05:05:13 +00:00
|
|
|
AST_LIST_TRAVERSE(&backends, chan, list) {
|
2005-03-04 06:47:24 +00:00
|
|
|
if (!strcasecmp(tech->type, chan->tech->type)) {
|
|
|
|
ast_log(LOG_WARNING, "Already have a handler for type '%s'\n", tech->type);
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
1999-12-04 21:35:07 +00:00
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
}
|
2006-02-15 01:48:54 +00:00
|
|
|
|
2006-04-19 06:58:20 +00:00
|
|
|
if (!(chan = ast_calloc(1, sizeof(*chan)))) {
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
1999-12-04 21:35:07 +00:00
|
|
|
return -1;
|
|
|
|
}
|
2006-01-11 06:35:34 +00:00
|
|
|
chan->tech = tech;
|
2006-01-11 05:05:13 +00:00
|
|
|
AST_LIST_INSERT_HEAD(&backends, chan, list);
|
2005-03-04 06:47:24 +00:00
|
|
|
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Registered handler for '%s' (%s)\n", chan->tech->type, chan->tech->description);
|
2005-03-04 06:47:24 +00:00
|
|
|
|
2007-07-26 15:49:18 +00:00
|
|
|
ast_verb(2, "Registered channel type '%s' (%s)\n", chan->tech->type, chan->tech->description);
|
2005-03-04 06:47:24 +00:00
|
|
|
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
1999-12-04 21:35:07 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2007-02-24 20:29:41 +00:00
|
|
|
/*! \brief Unregister channel driver */
|
2005-07-08 21:14:34 +00:00
|
|
|
void ast_channel_unregister(const struct ast_channel_tech *tech)
|
|
|
|
{
|
2006-01-11 05:05:13 +00:00
|
|
|
struct chanlist *chan;
|
2005-07-08 21:14:34 +00:00
|
|
|
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Unregistering channel type '%s'\n", tech->type);
|
2005-07-08 21:14:34 +00:00
|
|
|
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_WRLOCK(&channels);
|
2005-07-08 21:14:34 +00:00
|
|
|
|
2006-01-11 05:05:13 +00:00
|
|
|
AST_LIST_TRAVERSE_SAFE_BEGIN(&backends, chan, list) {
|
2005-07-08 21:14:34 +00:00
|
|
|
if (chan->tech == tech) {
|
2007-11-08 05:28:47 +00:00
|
|
|
AST_LIST_REMOVE_CURRENT(list);
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(chan);
|
2007-07-26 15:49:18 +00:00
|
|
|
ast_verb(2, "Unregistered channel type '%s'\n", tech->type);
|
2006-01-11 05:05:13 +00:00
|
|
|
break;
|
2005-07-08 21:14:34 +00:00
|
|
|
}
|
|
|
|
}
|
2007-11-08 05:28:47 +00:00
|
|
|
AST_LIST_TRAVERSE_SAFE_END;
|
2005-07-08 21:14:34 +00:00
|
|
|
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
2005-07-08 21:14:34 +00:00
|
|
|
}
|
|
|
|
|
2007-02-24 20:29:41 +00:00
|
|
|
/*! \brief Get handle to channel driver based on name */
|
2005-07-08 21:14:34 +00:00
|
|
|
const struct ast_channel_tech *ast_get_channel_tech(const char *name)
|
|
|
|
{
|
|
|
|
struct chanlist *chanls;
|
2006-01-11 05:05:13 +00:00
|
|
|
const struct ast_channel_tech *ret = NULL;
|
2005-07-08 21:14:34 +00:00
|
|
|
|
2008-01-28 00:56:14 +00:00
|
|
|
AST_RWLIST_RDLOCK(&channels);
|
2005-07-08 21:14:34 +00:00
|
|
|
|
2006-01-11 05:05:13 +00:00
|
|
|
AST_LIST_TRAVERSE(&backends, chanls, list) {
|
|
|
|
if (!strcasecmp(name, chanls->tech->type)) {
|
|
|
|
ret = chanls->tech;
|
|
|
|
break;
|
|
|
|
}
|
2005-07-08 21:14:34 +00:00
|
|
|
}
|
|
|
|
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
2006-01-11 05:05:13 +00:00
|
|
|
|
|
|
|
return ret;
|
2005-07-08 21:14:34 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Gives the string form of a given hangup cause */
|
2005-07-05 17:16:17 +00:00
|
|
|
const char *ast_cause2str(int cause)
|
|
|
|
{
|
|
|
|
int x;
|
|
|
|
|
2008-01-28 04:43:14 +00:00
|
|
|
for (x = 0; x < ARRAY_LEN(causes); x++) {
|
2005-07-05 17:16:17 +00:00
|
|
|
if (causes[x].cause == cause)
|
|
|
|
return causes[x].desc;
|
2006-04-06 15:40:12 +00:00
|
|
|
}
|
2005-07-05 17:16:17 +00:00
|
|
|
|
|
|
|
return "Unknown";
|
|
|
|
}
|
|
|
|
|
2006-05-25 20:51:27 +00:00
|
|
|
/*! \brief Convert a symbolic hangup cause to number */
|
|
|
|
int ast_str2cause(const char *name)
|
|
|
|
{
|
|
|
|
int x;
|
|
|
|
|
2008-01-28 04:43:14 +00:00
|
|
|
for (x = 0; x < ARRAY_LEN(causes); x++)
|
|
|
|
if (!strncasecmp(causes[x].name, name, strlen(causes[x].name)))
|
2006-05-25 20:51:27 +00:00
|
|
|
return causes[x].cause;
|
|
|
|
|
|
|
|
return -1;
|
|
|
|
}
|
2006-07-21 22:36:25 +00:00
|
|
|
|
2006-12-15 04:03:42 +00:00
|
|
|
/*! \brief Gives the string form of a given channel state.
|
|
|
|
\note This function is not reentrant.
|
|
|
|
*/
|
|
|
|
const char *ast_state2str(enum ast_channel_state state)
|
2001-05-07 00:43:32 +00:00
|
|
|
{
|
2006-07-21 22:36:25 +00:00
|
|
|
char *buf;
|
|
|
|
|
2007-01-23 00:11:32 +00:00
|
|
|
switch (state) {
|
2001-05-07 00:43:32 +00:00
|
|
|
case AST_STATE_DOWN:
|
|
|
|
return "Down";
|
|
|
|
case AST_STATE_RESERVED:
|
|
|
|
return "Rsrvd";
|
|
|
|
case AST_STATE_OFFHOOK:
|
|
|
|
return "OffHook";
|
|
|
|
case AST_STATE_DIALING:
|
|
|
|
return "Dialing";
|
|
|
|
case AST_STATE_RING:
|
|
|
|
return "Ring";
|
|
|
|
case AST_STATE_RINGING:
|
|
|
|
return "Ringing";
|
|
|
|
case AST_STATE_UP:
|
|
|
|
return "Up";
|
|
|
|
case AST_STATE_BUSY:
|
|
|
|
return "Busy";
|
2006-08-05 20:58:47 +00:00
|
|
|
case AST_STATE_DIALING_OFFHOOK:
|
|
|
|
return "Dialing Offhook";
|
|
|
|
case AST_STATE_PRERING:
|
|
|
|
return "Pre-ring";
|
2001-05-07 00:43:32 +00:00
|
|
|
default:
|
Merge team/russell/ast_verbose_threadstorage
- instead of defining a free() wrapper in a bunch of files, define it as
ast_free() in utils.h and remove the copies from all the files.
- centralize and abstract the code used for doing thread storage. The code
lives in threadstorage.h, with one function being implemented in utils.c.
This new API includes generic thread storage as well as special functions
for handling thread local dynamic length string buffers.
- update ast_inet_ntoa() to use the new threadstorage API
- update ast_state2str() to use the new threadstorage API
- update ast_cli() to use the new threadstorage API
- Modify manager_event() to use thread storage. Instead of using a buffer of
4096 characters as the workspace for building the manager event, use a thread
local dynamic string. Now there is no length limitation on the length of the
body of a manager event.
- Significantly simplify the handling of ast_verbose() ...
- Instead of using a static char buffer and a lock to make sure only one
thread can be using ast_verbose() at a time, use a thread local dynamic
string as the workspace for preparing the verbose message. Instead of
locking around the entire function, the only locking done now is when the
message has been built and is being deliviered to the list of registered
verbose message handlers.
- This function was doing a strdup() on every message passed to it and
keeping a queue of the last 200 messages in memory. This has been
completely removed. The only place this was used was that if there were
any messages in the verbose queue when a verbose handler was registered,
all of the messages in the queue would be fed to it. So, I just made sure
that the console verbose handler and the network verbose handler (for
remote asterisk consoles) were registered before any verbose messages.
pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at
startup, but I didn't feel the performance hit of this message queue was
worth saving the initial verbose output for these very rarely used modules.
- I have removed the last three arguments to the verbose handlers, leaving
only the string itself because they aren't needed anymore. For example,
ast_verbose had some logic for telling the verbose handler to add
a newline if the buffer was completely full. Now that the buffer can grow
as needed, this doesn't matter anymore.
- remove unused function, ast_verbose_dmesg() which was to dispatch the
message queue
- Convert the list of verbose handlers to use the linked list macros.
- add missing newline characters to a few ast_verbose() calls
- convert the list of log channels to use the linked list macros in logger.c
- fix close_logger() to close all of the files it opened for logging
- update ast_log() to use a thread local dynamic string for its workspace
for preparing log messages instead of a buffer of size BUFSIZ (8kB on my
system) allocated on the stack. The dynamic string in this case is limited
to only growing to a maximum size of BUFSIZ.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39272 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-08 06:32:04 +00:00
|
|
|
if (!(buf = ast_threadstorage_get(&state2str_threadbuf, STATE2STR_BUFSIZE)))
|
|
|
|
return "Unknown";
|
2006-08-05 20:58:47 +00:00
|
|
|
snprintf(buf, STATE2STR_BUFSIZE, "Unknown (%d)", state);
|
2006-07-21 22:36:25 +00:00
|
|
|
return buf;
|
2001-05-07 00:43:32 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Gives the string form of a given transfer capability */
|
2005-04-01 17:00:50 +00:00
|
|
|
char *ast_transfercapability2str(int transfercapability)
|
|
|
|
{
|
2007-01-23 00:11:32 +00:00
|
|
|
switch (transfercapability) {
|
2005-04-01 17:00:50 +00:00
|
|
|
case AST_TRANS_CAP_SPEECH:
|
|
|
|
return "SPEECH";
|
|
|
|
case AST_TRANS_CAP_DIGITAL:
|
|
|
|
return "DIGITAL";
|
|
|
|
case AST_TRANS_CAP_RESTRICTED_DIGITAL:
|
|
|
|
return "RESTRICTED_DIGITAL";
|
|
|
|
case AST_TRANS_CAP_3_1K_AUDIO:
|
|
|
|
return "3K1AUDIO";
|
|
|
|
case AST_TRANS_CAP_DIGITAL_W_TONES:
|
|
|
|
return "DIGITAL_W_TONES";
|
|
|
|
case AST_TRANS_CAP_VIDEO:
|
|
|
|
return "VIDEO";
|
|
|
|
default:
|
|
|
|
return "UNKNOWN";
|
|
|
|
}
|
|
|
|
}
|
2001-09-19 14:40:16 +00:00
|
|
|
|
2006-06-06 16:09:33 +00:00
|
|
|
/*! \brief Pick the best audio codec */
|
2001-09-19 14:40:16 +00:00
|
|
|
int ast_best_codec(int fmts)
|
|
|
|
{
|
|
|
|
/* This just our opinion, expressed in code. We are asked to choose
|
|
|
|
the best codec to use, given no information */
|
|
|
|
int x;
|
2007-03-01 20:24:59 +00:00
|
|
|
static const int prefs[] =
|
2001-09-19 14:40:16 +00:00
|
|
|
{
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! Okay, ulaw is used by all telephony equipment, so start with it */
|
2001-09-19 14:40:16 +00:00
|
|
|
AST_FORMAT_ULAW,
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! Unless of course, you're a silly European, so then prefer ALAW */
|
2001-09-19 14:40:16 +00:00
|
|
|
AST_FORMAT_ALAW,
|
2007-02-14 01:12:21 +00:00
|
|
|
/*! G.722 is better then all below, but not as common as the above... so give ulaw and alaw priority */
|
|
|
|
AST_FORMAT_G722,
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! Okay, well, signed linear is easy to translate into other stuff */
|
2008-01-17 23:20:37 +00:00
|
|
|
AST_FORMAT_SLINEAR16,
|
2001-09-19 14:40:16 +00:00
|
|
|
AST_FORMAT_SLINEAR,
|
2006-07-12 22:42:13 +00:00
|
|
|
/*! G.726 is standard ADPCM, in RFC3551 packing order */
|
2003-11-04 02:40:09 +00:00
|
|
|
AST_FORMAT_G726,
|
2006-07-12 22:42:13 +00:00
|
|
|
/*! G.726 is standard ADPCM, in AAL2 packing order */
|
|
|
|
AST_FORMAT_G726_AAL2,
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! ADPCM has great sound quality and is still pretty easy to translate */
|
2001-09-19 14:40:16 +00:00
|
|
|
AST_FORMAT_ADPCM,
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! Okay, we're down to vocoders now, so pick GSM because it's small and easier to
|
|
|
|
translate and sounds pretty good */
|
2001-09-19 14:40:16 +00:00
|
|
|
AST_FORMAT_GSM,
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! iLBC is not too bad */
|
2003-04-15 04:36:52 +00:00
|
|
|
AST_FORMAT_ILBC,
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! Speex is free, but computationally more expensive than GSM */
|
2002-09-10 20:48:20 +00:00
|
|
|
AST_FORMAT_SPEEX,
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! Ick, LPC10 sounds terrible, but at least we have code for it, if you're tacky enough
|
|
|
|
to use it */
|
2001-09-19 14:40:16 +00:00
|
|
|
AST_FORMAT_LPC10,
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! G.729a is faster than 723 and slightly less expensive */
|
2002-09-10 20:48:20 +00:00
|
|
|
AST_FORMAT_G729A,
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! Down to G.723.1 which is proprietary but at least designed for voice */
|
2001-09-19 14:40:16 +00:00
|
|
|
AST_FORMAT_G723_1,
|
|
|
|
};
|
2006-06-06 16:09:33 +00:00
|
|
|
|
|
|
|
/* Strip out video */
|
|
|
|
fmts &= AST_FORMAT_AUDIO_MASK;
|
2001-09-19 14:40:16 +00:00
|
|
|
|
2006-01-16 17:37:44 +00:00
|
|
|
/* Find the first preferred codec in the format given */
|
2008-07-08 16:40:28 +00:00
|
|
|
for (x = 0; x < ARRAY_LEN(prefs); x++) {
|
2001-09-19 14:40:16 +00:00
|
|
|
if (fmts & prefs[x])
|
|
|
|
return prefs[x];
|
2008-01-24 03:34:57 +00:00
|
|
|
}
|
|
|
|
|
2001-09-19 14:40:16 +00:00
|
|
|
ast_log(LOG_WARNING, "Don't know any of 0x%x formats\n", fmts);
|
2008-01-24 03:34:57 +00:00
|
|
|
|
2001-09-19 14:40:16 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2005-03-04 06:47:24 +00:00
|
|
|
static const struct ast_channel_tech null_tech = {
|
|
|
|
.type = "NULL",
|
2005-03-27 21:58:10 +00:00
|
|
|
.description = "Null channel (should not see this)",
|
2005-03-04 06:47:24 +00:00
|
|
|
};
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Create a new channel structure */
|
2007-04-10 05:41:34 +00:00
|
|
|
struct ast_channel *ast_channel_alloc(int needqueue, int state, const char *cid_num, const char *cid_name, const char *acctcode, const char *exten, const char *context, const int amaflag, const char *name_fmt, ...)
|
1999-12-04 21:35:07 +00:00
|
|
|
{
|
|
|
|
struct ast_channel *tmp;
|
2001-05-07 00:43:32 +00:00
|
|
|
int x;
|
2002-06-21 01:40:13 +00:00
|
|
|
int flags;
|
2006-04-06 15:40:12 +00:00
|
|
|
struct varshead *headp;
|
2006-11-07 21:47:49 +00:00
|
|
|
va_list ap1, ap2;
|
2005-03-04 06:47:24 +00:00
|
|
|
|
2002-06-21 01:40:13 +00:00
|
|
|
/* If shutting down, don't allocate any new channels */
|
2005-05-15 04:48:30 +00:00
|
|
|
if (shutting_down) {
|
|
|
|
ast_log(LOG_WARNING, "Channel allocation failed: Refusing due to active shutdown\n");
|
2002-06-21 01:40:13 +00:00
|
|
|
return NULL;
|
2005-05-15 04:48:30 +00:00
|
|
|
}
|
2005-03-04 06:47:24 +00:00
|
|
|
|
2006-04-06 15:40:12 +00:00
|
|
|
if (!(tmp = ast_calloc(1, sizeof(*tmp))))
|
2005-03-04 06:47:24 +00:00
|
|
|
return NULL;
|
|
|
|
|
2006-02-15 01:48:54 +00:00
|
|
|
if (!(tmp->sched = sched_context_create())) {
|
2005-05-15 04:48:30 +00:00
|
|
|
ast_log(LOG_WARNING, "Channel allocation failed: Unable to create schedule context\n");
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(tmp);
|
2005-03-04 06:47:24 +00:00
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
2007-05-24 22:25:55 +00:00
|
|
|
if ((ast_string_field_init(tmp, 128))) {
|
|
|
|
sched_context_destroy(tmp->sched);
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(tmp);
|
2007-05-24 22:08:33 +00:00
|
|
|
return NULL;
|
2007-05-24 22:25:55 +00:00
|
|
|
}
|
2006-02-01 23:05:28 +00:00
|
|
|
|
2007-08-08 21:44:58 +00:00
|
|
|
#ifdef HAVE_EPOLL
|
|
|
|
tmp->epfd = epoll_create(25);
|
|
|
|
#endif
|
|
|
|
|
|
|
|
for (x = 0; x < AST_MAX_FDS; x++) {
|
2005-03-04 06:47:24 +00:00
|
|
|
tmp->fds[x] = -1;
|
2007-08-08 21:44:58 +00:00
|
|
|
#ifdef HAVE_EPOLL
|
|
|
|
tmp->epfd_data[x] = NULL;
|
|
|
|
#endif
|
|
|
|
}
|
2005-03-04 06:47:24 +00:00
|
|
|
|
2008-06-13 12:45:50 +00:00
|
|
|
tmp->timingfd = ast_timer_open();
|
2005-03-04 06:47:24 +00:00
|
|
|
if (tmp->timingfd > -1) {
|
2008-06-13 12:45:50 +00:00
|
|
|
needqueue = 0;
|
2005-03-04 06:47:24 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
if (needqueue) {
|
|
|
|
if (pipe(tmp->alertpipe)) {
|
2005-05-15 04:48:30 +00:00
|
|
|
ast_log(LOG_WARNING, "Channel allocation failed: Can't create alert pipe!\n");
|
2008-06-12 15:14:37 +00:00
|
|
|
alertpipe_failed:
|
2008-06-13 12:45:50 +00:00
|
|
|
if (tmp->timingfd > -1) {
|
|
|
|
ast_timer_close(tmp->timingfd);
|
|
|
|
}
|
|
|
|
|
2007-11-26 15:36:27 +00:00
|
|
|
sched_context_destroy(tmp->sched);
|
2007-11-04 19:44:31 +00:00
|
|
|
ast_string_field_free_memory(tmp);
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(tmp);
|
2005-03-04 06:47:24 +00:00
|
|
|
return NULL;
|
|
|
|
} else {
|
|
|
|
flags = fcntl(tmp->alertpipe[0], F_GETFL);
|
2008-06-12 15:14:37 +00:00
|
|
|
if (fcntl(tmp->alertpipe[0], F_SETFL, flags | O_NONBLOCK) < 0) {
|
|
|
|
ast_log(LOG_WARNING, "Channel allocation failed: Unable to set alertpipe nonblocking! (%d: %s)\n", errno, strerror(errno));
|
|
|
|
close(tmp->alertpipe[0]);
|
|
|
|
close(tmp->alertpipe[1]);
|
|
|
|
goto alertpipe_failed;
|
|
|
|
}
|
2005-03-04 06:47:24 +00:00
|
|
|
flags = fcntl(tmp->alertpipe[1], F_GETFL);
|
2008-06-12 15:14:37 +00:00
|
|
|
if (fcntl(tmp->alertpipe[1], F_SETFL, flags | O_NONBLOCK) < 0) {
|
|
|
|
ast_log(LOG_WARNING, "Channel allocation failed: Unable to set alertpipe nonblocking! (%d: %s)\n", errno, strerror(errno));
|
|
|
|
close(tmp->alertpipe[0]);
|
|
|
|
close(tmp->alertpipe[1]);
|
|
|
|
goto alertpipe_failed;
|
|
|
|
}
|
1999-12-04 21:35:07 +00:00
|
|
|
}
|
2006-04-06 15:40:12 +00:00
|
|
|
} else /* Make sure we've got it done right if they don't */
|
2005-03-04 06:47:24 +00:00
|
|
|
tmp->alertpipe[0] = tmp->alertpipe[1] = -1;
|
|
|
|
|
|
|
|
/* Always watch the alertpipe */
|
2007-08-08 21:44:58 +00:00
|
|
|
ast_channel_set_fd(tmp, AST_ALERT_FD, tmp->alertpipe[0]);
|
2005-03-04 06:47:24 +00:00
|
|
|
/* And timing pipe */
|
2007-08-08 21:44:58 +00:00
|
|
|
ast_channel_set_fd(tmp, AST_TIMING_FD, tmp->timingfd);
|
2006-02-01 23:05:28 +00:00
|
|
|
ast_string_field_set(tmp, name, "**Unknown**");
|
2006-11-07 21:47:49 +00:00
|
|
|
|
2005-03-04 06:47:24 +00:00
|
|
|
/* Initial state */
|
2006-11-07 21:47:49 +00:00
|
|
|
tmp->_state = state;
|
|
|
|
|
2005-03-04 06:47:24 +00:00
|
|
|
tmp->streamid = -1;
|
2006-08-21 19:35:28 +00:00
|
|
|
|
2005-03-04 06:47:24 +00:00
|
|
|
tmp->fin = global_fin;
|
|
|
|
tmp->fout = global_fout;
|
2006-08-21 19:35:28 +00:00
|
|
|
|
|
|
|
if (ast_strlen_zero(ast_config_AST_SYSTEM_NAME)) {
|
|
|
|
ast_string_field_build(tmp, uniqueid, "%li.%d", (long) time(NULL),
|
2006-12-30 13:26:43 +00:00
|
|
|
ast_atomic_fetchadd_int(&uniqueint, 1));
|
2006-08-21 19:35:28 +00:00
|
|
|
} else {
|
|
|
|
ast_string_field_build(tmp, uniqueid, "%s-%li.%d", ast_config_AST_SYSTEM_NAME,
|
2006-12-30 13:26:43 +00:00
|
|
|
(long) time(NULL), ast_atomic_fetchadd_int(&uniqueint, 1));
|
2006-08-21 19:35:28 +00:00
|
|
|
}
|
|
|
|
|
2007-05-02 17:24:03 +00:00
|
|
|
tmp->cid.cid_name = ast_strdup(cid_name);
|
|
|
|
tmp->cid.cid_num = ast_strdup(cid_num);
|
|
|
|
|
2006-11-07 21:47:49 +00:00
|
|
|
if (!ast_strlen_zero(name_fmt)) {
|
|
|
|
/* Almost every channel is calling this function, and setting the name via the ast_string_field_build() call.
|
|
|
|
* And they all use slightly different formats for their name string.
|
|
|
|
* This means, to set the name here, we have to accept variable args, and call the string_field_build from here.
|
|
|
|
* This means, that the stringfields must have a routine that takes the va_lists directly, and
|
|
|
|
* uses them to build the string, instead of forming the va_lists internally from the vararg ... list.
|
|
|
|
* This new function was written so this can be accomplished.
|
|
|
|
*/
|
|
|
|
va_start(ap1, name_fmt);
|
|
|
|
va_start(ap2, name_fmt);
|
|
|
|
ast_string_field_build_va(tmp, name, name_fmt, ap1, ap2);
|
|
|
|
va_end(ap1);
|
|
|
|
va_end(ap2);
|
|
|
|
}
|
2007-03-30 14:37:21 +00:00
|
|
|
|
2007-03-30 17:57:47 +00:00
|
|
|
/* Reminder for the future: under what conditions do we NOT want to track cdrs on channels? */
|
2007-04-10 05:41:34 +00:00
|
|
|
|
|
|
|
/* These 4 variables need to be set up for the cdr_init() to work right */
|
|
|
|
if (amaflag)
|
|
|
|
tmp->amaflags = amaflag;
|
|
|
|
else
|
|
|
|
tmp->amaflags = ast_default_amaflags;
|
|
|
|
|
|
|
|
if (!ast_strlen_zero(acctcode))
|
|
|
|
ast_string_field_set(tmp, accountcode, acctcode);
|
|
|
|
else
|
|
|
|
ast_string_field_set(tmp, accountcode, ast_default_accountcode);
|
|
|
|
|
|
|
|
if (!ast_strlen_zero(context))
|
|
|
|
ast_copy_string(tmp->context, context, sizeof(tmp->context));
|
|
|
|
else
|
|
|
|
strcpy(tmp->context, "default");
|
|
|
|
|
|
|
|
if (!ast_strlen_zero(exten))
|
|
|
|
ast_copy_string(tmp->exten, exten, sizeof(tmp->exten));
|
|
|
|
else
|
|
|
|
strcpy(tmp->exten, "s");
|
|
|
|
|
|
|
|
tmp->priority = 1;
|
|
|
|
|
2007-03-30 14:37:21 +00:00
|
|
|
tmp->cdr = ast_cdr_alloc();
|
|
|
|
ast_cdr_init(tmp->cdr, tmp);
|
|
|
|
ast_cdr_start(tmp->cdr);
|
2006-11-07 21:47:49 +00:00
|
|
|
|
2005-03-04 06:47:24 +00:00
|
|
|
headp = &tmp->varshead;
|
2005-10-31 15:34:11 +00:00
|
|
|
AST_LIST_HEAD_INIT_NOLOCK(headp);
|
2006-08-21 19:35:28 +00:00
|
|
|
|
2007-11-15 16:20:47 +00:00
|
|
|
ast_mutex_init(&tmp->lock_dont_use);
|
2006-08-21 19:35:28 +00:00
|
|
|
|
2006-04-10 23:29:50 +00:00
|
|
|
AST_LIST_HEAD_INIT_NOLOCK(&tmp->datastores);
|
2006-08-21 19:35:28 +00:00
|
|
|
|
|
|
|
ast_string_field_set(tmp, language, defaultlanguage);
|
2005-03-04 06:47:24 +00:00
|
|
|
|
|
|
|
tmp->tech = &null_tech;
|
|
|
|
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_WRLOCK(&channels);
|
|
|
|
AST_RWLIST_INSERT_HEAD(&channels, tmp, chan_list);
|
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
2006-08-21 19:35:28 +00:00
|
|
|
|
2007-12-24 19:43:36 +00:00
|
|
|
/*\!note
|
|
|
|
* and now, since the channel structure is built, and has its name, let's
|
|
|
|
* call the manager event generator with this Newchannel event. This is the
|
|
|
|
* proper and correct place to make this call, but you sure do have to pass
|
|
|
|
* a lot of data into this func to do it here!
|
|
|
|
*/
|
|
|
|
if (!ast_strlen_zero(name_fmt)) {
|
|
|
|
manager_event(EVENT_FLAG_CALL, "Newchannel",
|
|
|
|
"Channel: %s\r\n"
|
|
|
|
"ChannelState: %d\r\n"
|
|
|
|
"ChannelStateDesc: %s\r\n"
|
|
|
|
"CallerIDNum: %s\r\n"
|
|
|
|
"CallerIDName: %s\r\n"
|
|
|
|
"AccountCode: %s\r\n"
|
|
|
|
"Uniqueid: %s\r\n",
|
|
|
|
tmp->name,
|
|
|
|
state,
|
|
|
|
ast_state2str(state),
|
|
|
|
S_OR(cid_num, ""),
|
|
|
|
S_OR(cid_name, ""),
|
|
|
|
tmp->accountcode,
|
|
|
|
tmp->uniqueid);
|
|
|
|
}
|
|
|
|
|
1999-12-04 21:35:07 +00:00
|
|
|
return tmp;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Queue an outgoing media frame */
|
2004-04-06 22:17:32 +00:00
|
|
|
int ast_queue_frame(struct ast_channel *chan, struct ast_frame *fin)
|
2002-06-21 01:40:13 +00:00
|
|
|
{
|
|
|
|
struct ast_frame *f;
|
2006-08-29 20:50:36 +00:00
|
|
|
struct ast_frame *cur;
|
2002-06-21 01:40:13 +00:00
|
|
|
int blah = 1;
|
|
|
|
int qlen = 0;
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2003-02-05 21:18:09 +00:00
|
|
|
/* Build us a copy and free the original one */
|
2006-02-15 01:48:54 +00:00
|
|
|
if (!(f = ast_frdup(fin))) {
|
2002-06-21 01:40:13 +00:00
|
|
|
ast_log(LOG_WARNING, "Unable to duplicate frame\n");
|
|
|
|
return -1;
|
|
|
|
}
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_lock(chan);
|
2006-08-29 20:50:36 +00:00
|
|
|
|
|
|
|
/* See if the last frame on the queue is a hangup, if so don't queue anything */
|
|
|
|
if ((cur = AST_LIST_LAST(&chan->readq)) && (cur->frametype == AST_FRAME_CONTROL) && (cur->subclass == AST_CONTROL_HANGUP)) {
|
|
|
|
ast_frfree(f);
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Count how many frames exist on the queue */
|
|
|
|
AST_LIST_TRAVERSE(&chan->readq, cur, frame_list) {
|
2002-06-21 01:40:13 +00:00
|
|
|
qlen++;
|
|
|
|
}
|
2006-08-29 20:50:36 +00:00
|
|
|
|
2003-03-18 06:00:18 +00:00
|
|
|
/* Allow up to 96 voice frames outstanding, and up to 128 total frames */
|
|
|
|
if (((fin->frametype == AST_FRAME_VOICE) && (qlen > 96)) || (qlen > 128)) {
|
2003-02-05 21:18:09 +00:00
|
|
|
if (fin->frametype != AST_FRAME_VOICE) {
|
|
|
|
ast_log(LOG_WARNING, "Exceptionally long queue length queuing to %s\n", chan->name);
|
2008-06-11 18:19:24 +00:00
|
|
|
ast_assert(fin->frametype == AST_FRAME_VOICE);
|
2003-02-05 21:18:09 +00:00
|
|
|
} else {
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Dropping voice to exceptionally long queue on %s\n", chan->name);
|
2003-05-08 04:40:35 +00:00
|
|
|
ast_frfree(f);
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_unlock(chan);
|
2003-02-05 21:18:09 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
}
|
2006-08-29 20:50:36 +00:00
|
|
|
AST_LIST_INSERT_TAIL(&chan->readq, f, frame_list);
|
2005-03-04 06:47:24 +00:00
|
|
|
if (chan->alertpipe[1] > -1) {
|
|
|
|
if (write(chan->alertpipe[1], &blah, sizeof(blah)) != sizeof(blah))
|
2002-09-10 20:48:20 +00:00
|
|
|
ast_log(LOG_WARNING, "Unable to write to alert pipe on %s, frametype/subclass %d/%d (qlen = %d): %s!\n",
|
|
|
|
chan->name, f->frametype, f->subclass, qlen, strerror(errno));
|
2004-03-19 04:23:20 +00:00
|
|
|
} else if (chan->timingfd > -1) {
|
2008-06-13 12:45:50 +00:00
|
|
|
ast_timer_enable_continuous(chan->timingfd);
|
2004-12-07 20:38:43 +00:00
|
|
|
} else if (ast_test_flag(chan, AST_FLAG_BLOCKING)) {
|
2003-04-11 04:31:33 +00:00
|
|
|
pthread_kill(chan->blocker, SIGURG);
|
2002-06-21 01:40:13 +00:00
|
|
|
}
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_unlock(chan);
|
2002-06-21 01:40:13 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Queue a hangup frame for channel */
|
2008-05-22 16:29:54 +00:00
|
|
|
int ast_queue_hangup(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_HANGUP };
|
|
|
|
/* Yeah, let's not change a lock-critical value without locking */
|
|
|
|
if (!ast_channel_trylock(chan)) {
|
|
|
|
chan->_softhangup |= AST_SOFTHANGUP_DEV;
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
}
|
|
|
|
return ast_queue_frame(chan, &f);
|
|
|
|
}
|
|
|
|
|
|
|
|
/*! \brief Queue a hangup frame for channel */
|
|
|
|
int ast_queue_hangup_with_cause(struct ast_channel *chan, int cause)
|
2002-06-21 01:40:13 +00:00
|
|
|
{
|
|
|
|
struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_HANGUP };
|
2008-04-24 22:16:48 +00:00
|
|
|
|
|
|
|
if (cause >= 0)
|
2008-05-22 16:29:54 +00:00
|
|
|
f.data.uint32 = cause;
|
2008-04-24 22:16:48 +00:00
|
|
|
|
2005-11-16 18:11:28 +00:00
|
|
|
/* Yeah, let's not change a lock-critical value without locking */
|
2006-04-26 21:44:25 +00:00
|
|
|
if (!ast_channel_trylock(chan)) {
|
2005-11-16 18:11:28 +00:00
|
|
|
chan->_softhangup |= AST_SOFTHANGUP_DEV;
|
2008-04-24 22:16:48 +00:00
|
|
|
if (cause < 0)
|
2008-05-22 16:29:54 +00:00
|
|
|
f.data.uint32 = chan->hangupcause;
|
2008-04-24 22:16:48 +00:00
|
|
|
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_unlock(chan);
|
2005-11-16 18:11:28 +00:00
|
|
|
}
|
2008-06-09 16:53:26 +00:00
|
|
|
|
2004-04-06 22:17:32 +00:00
|
|
|
return ast_queue_frame(chan, &f);
|
2002-06-21 01:40:13 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Queue a control frame */
|
2006-05-09 14:25:57 +00:00
|
|
|
int ast_queue_control(struct ast_channel *chan, enum ast_control_frame_type control)
|
2002-06-21 01:40:13 +00:00
|
|
|
{
|
|
|
|
struct ast_frame f = { AST_FRAME_CONTROL, };
|
2006-05-09 14:25:57 +00:00
|
|
|
|
|
|
|
f.subclass = control;
|
|
|
|
|
|
|
|
return ast_queue_frame(chan, &f);
|
|
|
|
}
|
|
|
|
|
|
|
|
/*! \brief Queue a control frame with payload */
|
|
|
|
int ast_queue_control_data(struct ast_channel *chan, enum ast_control_frame_type control,
|
|
|
|
const void *data, size_t datalen)
|
|
|
|
{
|
|
|
|
struct ast_frame f = { AST_FRAME_CONTROL, };
|
|
|
|
|
2002-06-21 01:40:13 +00:00
|
|
|
f.subclass = control;
|
2008-05-22 16:29:54 +00:00
|
|
|
f.data.ptr = (void *) data;
|
2006-05-09 14:25:57 +00:00
|
|
|
f.datalen = datalen;
|
|
|
|
|
2004-04-06 22:17:32 +00:00
|
|
|
return ast_queue_frame(chan, &f);
|
2002-06-21 01:40:13 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Set defer DTMF flag on channel */
|
2001-09-19 14:40:16 +00:00
|
|
|
int ast_channel_defer_dtmf(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
int pre = 0;
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2001-09-19 14:40:16 +00:00
|
|
|
if (chan) {
|
2004-12-07 20:38:43 +00:00
|
|
|
pre = ast_test_flag(chan, AST_FLAG_DEFER_DTMF);
|
|
|
|
ast_set_flag(chan, AST_FLAG_DEFER_DTMF);
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
|
|
|
return pre;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Unset defer DTMF flag on channel */
|
2001-09-19 14:40:16 +00:00
|
|
|
void ast_channel_undefer_dtmf(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
if (chan)
|
2004-12-07 20:38:43 +00:00
|
|
|
ast_clear_flag(chan, AST_FLAG_DEFER_DTMF);
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*!
|
2006-04-06 15:40:12 +00:00
|
|
|
* \brief Helper function to find channels.
|
|
|
|
*
|
2005-12-31 16:35:48 +00:00
|
|
|
* It supports these modes:
|
2005-09-15 16:34:39 +00:00
|
|
|
*
|
|
|
|
* prev != NULL : get channel next in list after prev
|
|
|
|
* name != NULL : get channel with matching name
|
|
|
|
* name != NULL && namelen != 0 : get channel whose name starts with prefix
|
|
|
|
* exten != NULL : get channel whose exten or macroexten matches
|
|
|
|
* context != NULL && exten != NULL : get channel whose context or macrocontext
|
2006-04-06 15:40:12 +00:00
|
|
|
*
|
2005-09-15 16:34:39 +00:00
|
|
|
* It returns with the channel's lock held. If getting the individual lock fails,
|
2005-06-06 02:29:18 +00:00
|
|
|
* unlock and retry quickly up to 10 times, then give up.
|
2006-04-06 15:40:12 +00:00
|
|
|
*
|
2005-12-31 16:35:48 +00:00
|
|
|
* \note XXX Note that this code has cost O(N) because of the need to verify
|
2005-06-06 02:29:18 +00:00
|
|
|
* that the object is still on the global list.
|
|
|
|
*
|
2005-12-31 16:35:48 +00:00
|
|
|
* \note XXX also note that accessing fields (e.g. c->name in ast_log())
|
2005-06-06 02:29:18 +00:00
|
|
|
* can only be done with the lock held or someone could delete the
|
|
|
|
* object while we work on it. This causes some ugliness in the code.
|
|
|
|
* Note that removing the first ast_log() may be harmful, as it would
|
|
|
|
* shorten the retry period and possibly cause failures.
|
|
|
|
* We should definitely go for a better scheme that is deadlock-free.
|
|
|
|
*/
|
|
|
|
static struct ast_channel *channel_find_locked(const struct ast_channel *prev,
|
2005-09-15 16:34:39 +00:00
|
|
|
const char *name, const int namelen,
|
|
|
|
const char *context, const char *exten)
|
1999-12-18 07:01:48 +00:00
|
|
|
{
|
2005-06-14 18:09:56 +00:00
|
|
|
const char *msg = prev ? "deadlock" : "initial deadlock";
|
2006-04-16 16:58:56 +00:00
|
|
|
int retries;
|
2005-06-06 02:29:18 +00:00
|
|
|
struct ast_channel *c;
|
2007-07-30 15:49:30 +00:00
|
|
|
const struct ast_channel *_prev = prev;
|
2005-06-06 02:29:18 +00:00
|
|
|
|
2008-04-14 17:42:20 +00:00
|
|
|
for (retries = 0; retries < 200; retries++) {
|
2006-04-16 16:58:56 +00:00
|
|
|
int done;
|
2008-08-06 03:55:49 +00:00
|
|
|
/* Reset prev on each retry. See note below for the reason. */
|
|
|
|
prev = _prev;
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_RDLOCK(&channels);
|
|
|
|
AST_RWLIST_TRAVERSE(&channels, c, chan_list) {
|
2008-08-06 03:55:49 +00:00
|
|
|
if (prev) { /* look for last item, first, before any evaluation */
|
2006-04-16 16:58:56 +00:00
|
|
|
if (c != prev) /* not this one */
|
|
|
|
continue;
|
|
|
|
/* found, prepare to return c->next */
|
2007-06-11 14:41:39 +00:00
|
|
|
if ((c = AST_RWLIST_NEXT(c, chan_list)) == NULL) break;
|
2008-08-06 03:55:49 +00:00
|
|
|
/*!\note
|
|
|
|
* We're done searching through the list for the previous item.
|
|
|
|
* Any item after this point, we want to evaluate for a match.
|
|
|
|
* If we didn't set prev to NULL here, then we would only
|
|
|
|
* return matches for the first matching item (since the above
|
|
|
|
* "if (c != prev)" would not permit any other potential
|
|
|
|
* matches to reach the additional matching logic, below).
|
|
|
|
* Instead, it would just iterate until it once again found the
|
|
|
|
* original match, then iterate down to the end of the list and
|
|
|
|
* quit.
|
2007-02-25 14:53:40 +00:00
|
|
|
*/
|
2007-06-07 18:41:17 +00:00
|
|
|
prev = NULL;
|
2006-07-28 23:36:06 +00:00
|
|
|
}
|
|
|
|
if (name) { /* want match by name */
|
2008-01-03 21:27:57 +00:00
|
|
|
if ((!namelen && strcasecmp(c->name, name) && strcmp(c->uniqueid, name)) ||
|
2006-07-28 23:36:06 +00:00
|
|
|
(namelen && strncasecmp(c->name, name, namelen)))
|
2006-04-16 16:58:56 +00:00
|
|
|
continue; /* name match failed */
|
|
|
|
} else if (exten) {
|
|
|
|
if (context && strcasecmp(c->context, context) &&
|
2006-07-28 23:36:06 +00:00
|
|
|
strcasecmp(c->macrocontext, context))
|
2006-04-16 16:58:56 +00:00
|
|
|
continue; /* context match failed */
|
|
|
|
if (strcasecmp(c->exten, exten) &&
|
2006-07-28 23:36:06 +00:00
|
|
|
strcasecmp(c->macroexten, exten))
|
2006-04-16 16:58:56 +00:00
|
|
|
continue; /* exten match failed */
|
2004-06-29 12:38:04 +00:00
|
|
|
}
|
2006-04-16 16:58:56 +00:00
|
|
|
/* if we get here, c points to the desired record */
|
|
|
|
break;
|
2004-06-29 12:38:04 +00:00
|
|
|
}
|
2005-06-06 02:29:18 +00:00
|
|
|
/* exit if chan not found or mutex acquired successfully */
|
2006-04-26 21:44:25 +00:00
|
|
|
/* this is slightly unsafe, as we _should_ hold the lock to access c->name */
|
|
|
|
done = c == NULL || ast_channel_trylock(c) == 0;
|
2006-10-03 15:53:07 +00:00
|
|
|
if (!done) {
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Avoiding %s for channel '%p'\n", msg, c);
|
2008-04-14 17:42:20 +00:00
|
|
|
if (retries == 199) {
|
2007-06-11 14:35:02 +00:00
|
|
|
/* We are about to fail due to a deadlock, so report this
|
|
|
|
* while we still have the list lock.
|
|
|
|
*/
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Failure, could not lock '%p' after %d retries!\n", c, retries);
|
2007-06-11 14:35:02 +00:00
|
|
|
/* As we have deadlocked, we will skip this channel and
|
|
|
|
* see if there is another match.
|
|
|
|
* NOTE: No point doing this for a full-name match,
|
|
|
|
* as there can be no more matches.
|
|
|
|
*/
|
|
|
|
if (!(name && !namelen)) {
|
|
|
|
prev = c;
|
|
|
|
retries = -1;
|
|
|
|
}
|
|
|
|
}
|
2006-10-03 15:53:07 +00:00
|
|
|
}
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
2005-06-06 02:29:18 +00:00
|
|
|
if (done)
|
|
|
|
return c;
|
2007-07-30 15:49:30 +00:00
|
|
|
/* If we reach this point we basically tried to lock a channel and failed. Instead of
|
|
|
|
* starting from the beginning of the list we can restore our saved pointer to the previous
|
|
|
|
* channel and start from there.
|
|
|
|
*/
|
|
|
|
prev = _prev;
|
2006-04-16 16:58:56 +00:00
|
|
|
usleep(1); /* give other threads a chance before retrying */
|
2004-06-29 10:51:00 +00:00
|
|
|
}
|
2005-06-06 02:29:18 +00:00
|
|
|
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Browse channels in use */
|
2005-06-06 02:29:18 +00:00
|
|
|
struct ast_channel *ast_channel_walk_locked(const struct ast_channel *prev)
|
|
|
|
{
|
2005-09-15 16:34:39 +00:00
|
|
|
return channel_find_locked(prev, NULL, 0, NULL, NULL);
|
1999-12-18 07:01:48 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Get channel by name and lock it */
|
2005-06-06 02:29:18 +00:00
|
|
|
struct ast_channel *ast_get_channel_by_name_locked(const char *name)
|
2004-05-22 04:11:22 +00:00
|
|
|
{
|
2005-09-15 16:34:39 +00:00
|
|
|
return channel_find_locked(NULL, name, 0, NULL, NULL);
|
2005-07-08 21:14:34 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Get channel by name prefix and lock it */
|
2005-07-08 21:14:34 +00:00
|
|
|
struct ast_channel *ast_get_channel_by_name_prefix_locked(const char *name, const int namelen)
|
|
|
|
{
|
2005-09-15 16:34:39 +00:00
|
|
|
return channel_find_locked(NULL, name, namelen, NULL, NULL);
|
2005-10-16 16:11:52 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Get next channel by name prefix and lock it */
|
2006-07-27 23:16:08 +00:00
|
|
|
struct ast_channel *ast_walk_channel_by_name_prefix_locked(const struct ast_channel *chan, const char *name,
|
|
|
|
const int namelen)
|
2005-10-16 16:11:52 +00:00
|
|
|
{
|
|
|
|
return channel_find_locked(chan, name, namelen, NULL, NULL);
|
2005-09-15 16:34:39 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Get channel by exten (and optionally context) and lock it */
|
2005-09-15 16:34:39 +00:00
|
|
|
struct ast_channel *ast_get_channel_by_exten_locked(const char *exten, const char *context)
|
|
|
|
{
|
|
|
|
return channel_find_locked(NULL, NULL, 0, context, exten);
|
2004-05-22 04:11:22 +00:00
|
|
|
}
|
|
|
|
|
2006-07-28 23:36:06 +00:00
|
|
|
/*! \brief Get next channel by exten (and optionally context) and lock it */
|
|
|
|
struct ast_channel *ast_walk_channel_by_exten_locked(const struct ast_channel *chan, const char *exten,
|
|
|
|
const char *context)
|
|
|
|
{
|
|
|
|
return channel_find_locked(chan, NULL, 0, context, exten);
|
|
|
|
}
|
|
|
|
|
2008-11-04 23:23:39 +00:00
|
|
|
/*! \brief Search for a channel based on the passed channel matching callback (first match) and return it, locked */
|
2008-11-09 01:59:59 +00:00
|
|
|
struct ast_channel *ast_channel_search_locked(int (*is_match)(struct ast_channel *, void *), void *data)
|
2008-11-04 23:23:39 +00:00
|
|
|
{
|
|
|
|
struct ast_channel *c = NULL;
|
|
|
|
|
|
|
|
AST_RWLIST_RDLOCK(&channels);
|
|
|
|
AST_RWLIST_TRAVERSE(&channels, c, chan_list) {
|
|
|
|
ast_channel_lock(c);
|
2008-11-09 01:59:59 +00:00
|
|
|
if (is_match(c, data)) {
|
2008-11-04 23:23:39 +00:00
|
|
|
break;
|
|
|
|
}
|
|
|
|
ast_channel_unlock(c);
|
|
|
|
}
|
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
|
|
|
|
|
|
|
return c;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Wait, look for hangups and condition arg */
|
2006-01-01 08:45:29 +00:00
|
|
|
int ast_safe_sleep_conditional(struct ast_channel *chan, int ms, int (*cond)(void*), void *data)
|
2003-02-05 21:18:09 +00:00
|
|
|
{
|
|
|
|
struct ast_frame *f;
|
|
|
|
|
2006-01-01 08:45:29 +00:00
|
|
|
while (ms > 0) {
|
|
|
|
if (cond && ((*cond)(data) == 0))
|
2003-02-05 21:18:09 +00:00
|
|
|
return 0;
|
|
|
|
ms = ast_waitfor(chan, ms);
|
2006-01-01 08:45:29 +00:00
|
|
|
if (ms < 0)
|
2003-02-05 21:18:09 +00:00
|
|
|
return -1;
|
|
|
|
if (ms > 0) {
|
|
|
|
f = ast_read(chan);
|
|
|
|
if (!f)
|
|
|
|
return -1;
|
|
|
|
ast_frfree(f);
|
|
|
|
}
|
|
|
|
}
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Wait, look for hangups */
|
2002-06-21 01:40:13 +00:00
|
|
|
int ast_safe_sleep(struct ast_channel *chan, int ms)
|
|
|
|
{
|
2006-01-01 08:45:29 +00:00
|
|
|
return ast_safe_sleep_conditional(chan, ms, NULL, NULL);
|
2002-06-21 01:40:13 +00:00
|
|
|
}
|
|
|
|
|
2004-10-02 00:58:31 +00:00
|
|
|
static void free_cid(struct ast_callerid *cid)
|
|
|
|
{
|
|
|
|
if (cid->cid_dnid)
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(cid->cid_dnid);
|
2004-10-02 00:58:31 +00:00
|
|
|
if (cid->cid_num)
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(cid->cid_num);
|
2004-10-02 00:58:31 +00:00
|
|
|
if (cid->cid_name)
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(cid->cid_name);
|
2004-10-02 00:58:31 +00:00
|
|
|
if (cid->cid_ani)
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(cid->cid_ani);
|
2004-10-02 00:58:31 +00:00
|
|
|
if (cid->cid_rdnis)
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(cid->cid_rdnis);
|
2006-11-25 09:02:42 +00:00
|
|
|
cid->cid_dnid = cid->cid_num = cid->cid_name = cid->cid_ani = cid->cid_rdnis = NULL;
|
2004-10-02 00:58:31 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Free a channel structure */
|
1999-12-18 07:01:48 +00:00
|
|
|
void ast_channel_free(struct ast_channel *chan)
|
|
|
|
{
|
2002-06-21 01:40:13 +00:00
|
|
|
int fd;
|
2007-08-08 21:44:58 +00:00
|
|
|
#ifdef HAVE_EPOLL
|
|
|
|
int i;
|
|
|
|
#endif
|
2002-09-10 20:48:20 +00:00
|
|
|
struct ast_var_t *vardata;
|
2006-08-29 20:50:36 +00:00
|
|
|
struct ast_frame *f;
|
2002-09-10 20:48:20 +00:00
|
|
|
struct varshead *headp;
|
2006-04-10 23:29:50 +00:00
|
|
|
struct ast_datastore *datastore = NULL;
|
2008-07-25 17:24:43 +00:00
|
|
|
char name[AST_CHANNEL_NAME], *dashptr;
|
2002-09-10 20:48:20 +00:00
|
|
|
|
|
|
|
headp=&chan->varshead;
|
|
|
|
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_WRLOCK(&channels);
|
|
|
|
if (!AST_RWLIST_REMOVE(&channels, chan, chan_list)) {
|
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
2007-06-06 16:58:28 +00:00
|
|
|
ast_log(LOG_ERROR, "Unable to find channel in list to free. Assuming it has already been done.\n");
|
|
|
|
}
|
2008-03-12 19:59:05 +00:00
|
|
|
/* Lock and unlock the channel just to be sure nobody has it locked still
|
|
|
|
due to a reference retrieved from the channel list. */
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_lock(chan);
|
|
|
|
ast_channel_unlock(chan);
|
2008-03-12 19:59:05 +00:00
|
|
|
|
|
|
|
/* Get rid of each of the data stores on the channel */
|
|
|
|
while ((datastore = AST_LIST_REMOVE_HEAD(&chan->datastores, entry)))
|
|
|
|
/* Free the data store */
|
2008-08-05 16:56:11 +00:00
|
|
|
ast_datastore_free(datastore);
|
2008-03-12 19:59:05 +00:00
|
|
|
|
|
|
|
/* Lock and unlock the channel just to be sure nobody has it locked still
|
|
|
|
due to a reference that was stored in a datastore. (i.e. app_chanspy) */
|
|
|
|
ast_channel_lock(chan);
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
|
2005-03-04 06:47:24 +00:00
|
|
|
if (chan->tech_pvt) {
|
1999-12-18 07:01:48 +00:00
|
|
|
ast_log(LOG_WARNING, "Channel '%s' may not have been hung up properly\n", chan->name);
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(chan->tech_pvt);
|
2005-03-04 06:47:24 +00:00
|
|
|
}
|
2003-03-30 22:55:42 +00:00
|
|
|
|
Merged revisions 7265-7266,7268-7275 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r7265 | oej | 2005-12-01 17:18:14 -0600 (Thu, 01 Dec 2005) | 2 lines
Changing bug report address to the Asterisk issue tracker
........
r7266 | kpfleming | 2005-12-01 17:18:29 -0600 (Thu, 01 Dec 2005) | 3 lines
Makefile 'update' target now supports updating from Subversion repositories (issue #5875)
remove support for 'patches' subdirectory, it's no longer useful
........
r7268 | kpfleming | 2005-12-01 17:34:58 -0600 (Thu, 01 Dec 2005) | 2 lines
ensure channel's scheduling context is freed (issue #5788)
........
r7269 | kpfleming | 2005-12-01 17:49:44 -0600 (Thu, 01 Dec 2005) | 2 lines
don't block waiting for the Festival server forever when it goes away (issue #5882)
........
r7270 | kpfleming | 2005-12-01 18:26:12 -0600 (Thu, 01 Dec 2005) | 2 lines
allow variables to exist on both 'halves' of the Local channel (issue #5810)
........
r7271 | kpfleming | 2005-12-01 18:28:48 -0600 (Thu, 01 Dec 2005) | 2 lines
protect agent_bridgedchannel() from segfaulting when there is no bridged channel (issue #5879)
........
r7272 | kpfleming | 2005-12-01 18:39:00 -0600 (Thu, 01 Dec 2005) | 3 lines
properly handle password changes when mailbox is last line of config file and not followed by a newline (issue #5870)
reformat password changing code to conform to coding guidelines (issue #5870)
........
r7273 | kpfleming | 2005-12-01 18:42:40 -0600 (Thu, 01 Dec 2005) | 2 lines
allow previous context-searching behavior to be used if desired (issue #5899)
........
r7274 | kpfleming | 2005-12-01 18:51:15 -0600 (Thu, 01 Dec 2005) | 2 lines
inherit channel variables into channels created by Page() application (issue #5888)
........
r7275 | oej | 2005-12-01 18:52:13 -0600 (Thu, 01 Dec 2005) | 2 lines
Bug #5907. Improve SIP INFO DTMF debugging output. (1.2 & Trunk)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7276 f38db490-d61c-443f-a65b-d21fe96a405b
2005-12-02 01:01:11 +00:00
|
|
|
if (chan->sched)
|
|
|
|
sched_context_destroy(chan->sched);
|
|
|
|
|
2005-06-05 16:32:16 +00:00
|
|
|
ast_copy_string(name, chan->name, sizeof(name));
|
2008-07-25 17:24:43 +00:00
|
|
|
if ((dashptr = strrchr(name, '-'))) {
|
|
|
|
*dashptr = '\0';
|
|
|
|
}
|
Merged revisions 7265-7266,7268-7275 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r7265 | oej | 2005-12-01 17:18:14 -0600 (Thu, 01 Dec 2005) | 2 lines
Changing bug report address to the Asterisk issue tracker
........
r7266 | kpfleming | 2005-12-01 17:18:29 -0600 (Thu, 01 Dec 2005) | 3 lines
Makefile 'update' target now supports updating from Subversion repositories (issue #5875)
remove support for 'patches' subdirectory, it's no longer useful
........
r7268 | kpfleming | 2005-12-01 17:34:58 -0600 (Thu, 01 Dec 2005) | 2 lines
ensure channel's scheduling context is freed (issue #5788)
........
r7269 | kpfleming | 2005-12-01 17:49:44 -0600 (Thu, 01 Dec 2005) | 2 lines
don't block waiting for the Festival server forever when it goes away (issue #5882)
........
r7270 | kpfleming | 2005-12-01 18:26:12 -0600 (Thu, 01 Dec 2005) | 2 lines
allow variables to exist on both 'halves' of the Local channel (issue #5810)
........
r7271 | kpfleming | 2005-12-01 18:28:48 -0600 (Thu, 01 Dec 2005) | 2 lines
protect agent_bridgedchannel() from segfaulting when there is no bridged channel (issue #5879)
........
r7272 | kpfleming | 2005-12-01 18:39:00 -0600 (Thu, 01 Dec 2005) | 3 lines
properly handle password changes when mailbox is last line of config file and not followed by a newline (issue #5870)
reformat password changing code to conform to coding guidelines (issue #5870)
........
r7273 | kpfleming | 2005-12-01 18:42:40 -0600 (Thu, 01 Dec 2005) | 2 lines
allow previous context-searching behavior to be used if desired (issue #5899)
........
r7274 | kpfleming | 2005-12-01 18:51:15 -0600 (Thu, 01 Dec 2005) | 2 lines
inherit channel variables into channels created by Page() application (issue #5888)
........
r7275 | oej | 2005-12-01 18:52:13 -0600 (Thu, 01 Dec 2005) | 2 lines
Bug #5907. Improve SIP INFO DTMF debugging output. (1.2 & Trunk)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7276 f38db490-d61c-443f-a65b-d21fe96a405b
2005-12-02 01:01:11 +00:00
|
|
|
|
2003-03-25 19:30:06 +00:00
|
|
|
/* Stop monitoring */
|
2006-07-28 18:59:59 +00:00
|
|
|
if (chan->monitor)
|
2003-03-25 19:30:06 +00:00
|
|
|
chan->monitor->stop( chan, 0 );
|
2003-03-30 22:55:42 +00:00
|
|
|
|
2004-12-24 01:40:07 +00:00
|
|
|
/* If there is native format music-on-hold state, free it */
|
2006-07-28 18:59:59 +00:00
|
|
|
if (chan->music_state)
|
2004-12-24 01:40:07 +00:00
|
|
|
ast_moh_cleanup(chan);
|
|
|
|
|
2006-01-16 17:37:44 +00:00
|
|
|
/* Free translators */
|
2005-03-04 06:47:24 +00:00
|
|
|
if (chan->readtrans)
|
|
|
|
ast_translator_free_path(chan->readtrans);
|
|
|
|
if (chan->writetrans)
|
|
|
|
ast_translator_free_path(chan->writetrans);
|
2006-04-06 15:40:12 +00:00
|
|
|
if (chan->pbx)
|
1999-12-18 07:01:48 +00:00
|
|
|
ast_log(LOG_WARNING, "PBX may not have been terminated properly on '%s'\n", chan->name);
|
2004-10-02 00:58:31 +00:00
|
|
|
free_cid(&chan->cid);
|
2002-06-21 01:40:13 +00:00
|
|
|
/* Close pipes if appropriate */
|
2005-03-04 06:47:24 +00:00
|
|
|
if ((fd = chan->alertpipe[0]) > -1)
|
2002-06-21 01:40:13 +00:00
|
|
|
close(fd);
|
2005-03-04 06:47:24 +00:00
|
|
|
if ((fd = chan->alertpipe[1]) > -1)
|
2002-06-21 01:40:13 +00:00
|
|
|
close(fd);
|
2003-07-04 16:49:11 +00:00
|
|
|
if ((fd = chan->timingfd) > -1)
|
2008-06-13 12:45:50 +00:00
|
|
|
ast_timer_close(fd);
|
2007-08-08 21:44:58 +00:00
|
|
|
#ifdef HAVE_EPOLL
|
|
|
|
for (i = 0; i < AST_MAX_FDS; i++) {
|
|
|
|
if (chan->epfd_data[i])
|
|
|
|
free(chan->epfd_data[i]);
|
|
|
|
}
|
|
|
|
close(chan->epfd);
|
|
|
|
#endif
|
2006-08-29 20:50:36 +00:00
|
|
|
while ((f = AST_LIST_REMOVE_HEAD(&chan->readq, frame_list)))
|
|
|
|
ast_frfree(f);
|
2002-09-10 20:48:20 +00:00
|
|
|
|
|
|
|
/* loop over the variables list, freeing all data and deleting list items */
|
|
|
|
/* no need to lock the list, as the channel is already locked */
|
|
|
|
|
2005-10-28 23:01:13 +00:00
|
|
|
while ((vardata = AST_LIST_REMOVE_HEAD(headp, entries)))
|
|
|
|
ast_var_delete(vardata);
|
2002-09-10 20:48:20 +00:00
|
|
|
|
2007-04-26 03:24:01 +00:00
|
|
|
ast_app_group_discard(chan);
|
|
|
|
|
2006-05-31 16:56:50 +00:00
|
|
|
/* Destroy the jitterbuffer */
|
2006-06-23 16:49:12 +00:00
|
|
|
ast_jb_destroy(chan);
|
2008-09-02 22:50:59 +00:00
|
|
|
|
|
|
|
if (chan->cdr) {
|
2008-09-05 14:18:43 +00:00
|
|
|
ast_cdr_discard(chan->cdr);
|
2008-09-02 22:50:59 +00:00
|
|
|
chan->cdr = NULL;
|
|
|
|
}
|
2008-03-12 17:02:57 +00:00
|
|
|
|
|
|
|
ast_mutex_destroy(&chan->lock_dont_use);
|
2006-05-31 16:56:50 +00:00
|
|
|
|
2007-11-04 19:44:31 +00:00
|
|
|
ast_string_field_free_memory(chan);
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(chan);
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
2003-03-30 22:55:42 +00:00
|
|
|
|
2008-07-25 21:20:03 +00:00
|
|
|
ast_devstate_changed_literal(AST_DEVICE_NOT_INUSE, name);
|
1999-12-18 07:01:48 +00:00
|
|
|
}
|
|
|
|
|
2007-08-28 18:32:56 +00:00
|
|
|
struct ast_datastore *ast_channel_datastore_alloc(const struct ast_datastore_info *info, const char *uid)
|
2006-04-10 23:29:50 +00:00
|
|
|
{
|
2008-08-05 16:56:11 +00:00
|
|
|
return ast_datastore_alloc(info, uid);
|
2006-04-10 23:29:50 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
int ast_channel_datastore_free(struct ast_datastore *datastore)
|
|
|
|
{
|
2008-08-05 16:56:11 +00:00
|
|
|
return ast_datastore_free(datastore);
|
2006-04-10 23:29:50 +00:00
|
|
|
}
|
|
|
|
|
2007-03-03 14:40:18 +00:00
|
|
|
int ast_channel_datastore_inherit(struct ast_channel *from, struct ast_channel *to)
|
|
|
|
{
|
|
|
|
struct ast_datastore *datastore = NULL, *datastore2;
|
|
|
|
|
|
|
|
AST_LIST_TRAVERSE(&from->datastores, datastore, entry) {
|
|
|
|
if (datastore->inheritance > 0) {
|
2008-08-05 16:56:11 +00:00
|
|
|
datastore2 = ast_datastore_alloc(datastore->info, datastore->uid);
|
2007-03-03 14:40:18 +00:00
|
|
|
if (datastore2) {
|
|
|
|
datastore2->data = datastore->info->duplicate(datastore->data);
|
|
|
|
datastore2->inheritance = datastore->inheritance == DATASTORE_INHERIT_FOREVER ? DATASTORE_INHERIT_FOREVER : datastore->inheritance - 1;
|
|
|
|
AST_LIST_INSERT_TAIL(&to->datastores, datastore2, entry);
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2006-04-10 23:29:50 +00:00
|
|
|
int ast_channel_datastore_add(struct ast_channel *chan, struct ast_datastore *datastore)
|
|
|
|
{
|
|
|
|
int res = 0;
|
|
|
|
|
2006-04-11 03:50:17 +00:00
|
|
|
AST_LIST_INSERT_HEAD(&chan->datastores, datastore, entry);
|
2006-04-10 23:29:50 +00:00
|
|
|
|
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
|
|
|
int ast_channel_datastore_remove(struct ast_channel *chan, struct ast_datastore *datastore)
|
|
|
|
{
|
2007-11-14 00:54:38 +00:00
|
|
|
return AST_LIST_REMOVE(&chan->datastores, datastore, entry) ? 0 : -1;
|
2006-04-10 23:29:50 +00:00
|
|
|
}
|
|
|
|
|
2007-08-28 18:32:56 +00:00
|
|
|
struct ast_datastore *ast_channel_datastore_find(struct ast_channel *chan, const struct ast_datastore_info *info, const char *uid)
|
2006-04-10 23:29:50 +00:00
|
|
|
{
|
|
|
|
struct ast_datastore *datastore = NULL;
|
|
|
|
|
|
|
|
if (info == NULL)
|
|
|
|
return NULL;
|
|
|
|
|
2006-04-11 03:50:17 +00:00
|
|
|
AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->datastores, datastore, entry) {
|
2008-08-05 16:56:11 +00:00
|
|
|
if (datastore->info != info) {
|
|
|
|
continue;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (uid == NULL) {
|
|
|
|
/* matched by type only */
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
|
|
|
|
if ((datastore->uid != NULL) && !strcasecmp(uid, datastore->uid)) {
|
|
|
|
/* Matched by type AND uid */
|
|
|
|
break;
|
2006-04-10 23:29:50 +00:00
|
|
|
}
|
|
|
|
}
|
2008-08-05 16:56:11 +00:00
|
|
|
AST_LIST_TRAVERSE_SAFE_END;
|
2006-04-10 23:29:50 +00:00
|
|
|
|
|
|
|
return datastore;
|
|
|
|
}
|
|
|
|
|
2007-08-08 21:44:58 +00:00
|
|
|
/*! Set the file descriptor on the channel */
|
|
|
|
void ast_channel_set_fd(struct ast_channel *chan, int which, int fd)
|
|
|
|
{
|
|
|
|
#ifdef HAVE_EPOLL
|
|
|
|
struct epoll_event ev;
|
|
|
|
struct ast_epoll_data *aed = NULL;
|
|
|
|
|
|
|
|
if (chan->fds[which] > -1) {
|
|
|
|
epoll_ctl(chan->epfd, EPOLL_CTL_DEL, chan->fds[which], &ev);
|
|
|
|
aed = chan->epfd_data[which];
|
|
|
|
}
|
|
|
|
|
|
|
|
/* If this new fd is valid, add it to the epoll */
|
|
|
|
if (fd > -1) {
|
|
|
|
if (!aed && (!(aed = ast_calloc(1, sizeof(*aed)))))
|
|
|
|
return;
|
|
|
|
|
|
|
|
chan->epfd_data[which] = aed;
|
|
|
|
aed->chan = chan;
|
|
|
|
aed->which = which;
|
|
|
|
|
|
|
|
ev.events = EPOLLIN | EPOLLPRI | EPOLLERR | EPOLLHUP;
|
|
|
|
ev.data.ptr = aed;
|
|
|
|
epoll_ctl(chan->epfd, EPOLL_CTL_ADD, fd, &ev);
|
|
|
|
} else if (aed) {
|
|
|
|
/* We don't have to keep around this epoll data structure now */
|
|
|
|
free(aed);
|
|
|
|
chan->epfd_data[which] = NULL;
|
|
|
|
}
|
|
|
|
#endif
|
|
|
|
chan->fds[which] = fd;
|
|
|
|
return;
|
|
|
|
}
|
|
|
|
|
|
|
|
/*! Add a channel to an optimized waitfor */
|
|
|
|
void ast_poll_channel_add(struct ast_channel *chan0, struct ast_channel *chan1)
|
|
|
|
{
|
|
|
|
#ifdef HAVE_EPOLL
|
|
|
|
struct epoll_event ev;
|
|
|
|
int i = 0;
|
|
|
|
|
|
|
|
if (chan0->epfd == -1)
|
|
|
|
return;
|
|
|
|
|
|
|
|
/* Iterate through the file descriptors on chan1, adding them to chan0 */
|
|
|
|
for (i = 0; i < AST_MAX_FDS; i++) {
|
|
|
|
if (chan1->fds[i] == -1)
|
|
|
|
continue;
|
|
|
|
ev.events = EPOLLIN | EPOLLPRI | EPOLLERR | EPOLLHUP;
|
|
|
|
ev.data.ptr = chan1->epfd_data[i];
|
|
|
|
epoll_ctl(chan0->epfd, EPOLL_CTL_ADD, chan1->fds[i], &ev);
|
|
|
|
}
|
|
|
|
|
|
|
|
#endif
|
|
|
|
return;
|
|
|
|
}
|
|
|
|
|
|
|
|
/*! Delete a channel from an optimized waitfor */
|
|
|
|
void ast_poll_channel_del(struct ast_channel *chan0, struct ast_channel *chan1)
|
|
|
|
{
|
|
|
|
#ifdef HAVE_EPOLL
|
|
|
|
struct epoll_event ev;
|
|
|
|
int i = 0;
|
|
|
|
|
|
|
|
if (chan0->epfd == -1)
|
|
|
|
return;
|
|
|
|
|
|
|
|
for (i = 0; i < AST_MAX_FDS; i++) {
|
|
|
|
if (chan1->fds[i] == -1)
|
|
|
|
continue;
|
|
|
|
epoll_ctl(chan0->epfd, EPOLL_CTL_DEL, chan1->fds[i], &ev);
|
|
|
|
}
|
|
|
|
|
|
|
|
#endif
|
|
|
|
return;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Softly hangup a channel, don't lock */
|
2002-09-10 20:48:20 +00:00
|
|
|
int ast_softhangup_nolock(struct ast_channel *chan, int cause)
|
1999-12-04 21:35:07 +00:00
|
|
|
{
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Soft-Hanging up channel '%s'\n", chan->name);
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Inform channel driver that we need to be hung up, if it cares */
|
2002-09-10 20:48:20 +00:00
|
|
|
chan->_softhangup |= cause;
|
2006-04-06 15:40:12 +00:00
|
|
|
ast_queue_frame(chan, &ast_null_frame);
|
2004-04-26 03:02:49 +00:00
|
|
|
/* Interrupt any poll call or such */
|
2004-12-07 20:38:43 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_BLOCKING))
|
1999-12-04 21:35:07 +00:00
|
|
|
pthread_kill(chan->blocker, SIGURG);
|
2006-04-06 15:40:12 +00:00
|
|
|
return 0;
|
1999-12-04 21:35:07 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Softly hangup a channel, lock */
|
2002-09-10 20:48:20 +00:00
|
|
|
int ast_softhangup(struct ast_channel *chan, int cause)
|
|
|
|
{
|
|
|
|
int res;
|
2008-06-09 16:48:26 +00:00
|
|
|
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_lock(chan);
|
2002-09-10 20:48:20 +00:00
|
|
|
res = ast_softhangup_nolock(chan, cause);
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_unlock(chan);
|
2008-06-09 16:48:26 +00:00
|
|
|
|
2002-09-10 20:48:20 +00:00
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
2008-08-10 19:35:50 +00:00
|
|
|
static void free_translation(struct ast_channel *clonechan)
|
2001-09-19 14:40:16 +00:00
|
|
|
{
|
2008-08-10 19:35:50 +00:00
|
|
|
if (clonechan->writetrans)
|
|
|
|
ast_translator_free_path(clonechan->writetrans);
|
|
|
|
if (clonechan->readtrans)
|
|
|
|
ast_translator_free_path(clonechan->readtrans);
|
|
|
|
clonechan->writetrans = NULL;
|
|
|
|
clonechan->readtrans = NULL;
|
|
|
|
clonechan->rawwriteformat = clonechan->nativeformats;
|
|
|
|
clonechan->rawreadformat = clonechan->nativeformats;
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Hangup a channel */
|
1999-12-04 21:35:07 +00:00
|
|
|
int ast_hangup(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
int res = 0;
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Don't actually hang up a channel that will masquerade as someone else, or
|
|
|
|
if someone is going to masquerade as us */
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_lock(chan);
|
2005-03-23 21:52:31 +00:00
|
|
|
|
2007-08-08 19:30:52 +00:00
|
|
|
if (chan->audiohooks) {
|
|
|
|
ast_audiohook_detach_list(chan->audiohooks);
|
|
|
|
chan->audiohooks = NULL;
|
|
|
|
}
|
2005-03-23 21:52:31 +00:00
|
|
|
|
2007-11-12 17:44:04 +00:00
|
|
|
ast_autoservice_stop(chan);
|
|
|
|
|
2001-09-19 14:40:16 +00:00
|
|
|
if (chan->masq) {
|
2006-04-06 15:40:12 +00:00
|
|
|
if (ast_do_masquerade(chan))
|
2003-02-05 21:18:09 +00:00
|
|
|
ast_log(LOG_WARNING, "Failed to perform masquerade\n");
|
|
|
|
}
|
|
|
|
|
|
|
|
if (chan->masq) {
|
|
|
|
ast_log(LOG_WARNING, "%s getting hung up, but someone is trying to masq into us?!?\n", chan->name);
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_unlock(chan);
|
2001-05-07 00:43:32 +00:00
|
|
|
return 0;
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
2006-04-06 15:40:12 +00:00
|
|
|
/* If this channel is one which will be masqueraded into something,
|
2001-05-07 00:43:32 +00:00
|
|
|
mark it as a zombie already, so we know to free it later */
|
|
|
|
if (chan->masqr) {
|
2004-12-07 20:38:43 +00:00
|
|
|
ast_set_flag(chan, AST_FLAG_ZOMBIE);
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_unlock(chan);
|
2001-05-07 00:43:32 +00:00
|
|
|
return 0;
|
|
|
|
}
|
2001-09-19 14:40:16 +00:00
|
|
|
free_translation(chan);
|
2006-08-07 00:17:12 +00:00
|
|
|
/* Close audio stream */
|
|
|
|
if (chan->stream) {
|
2004-04-02 23:25:24 +00:00
|
|
|
ast_closestream(chan->stream);
|
2006-08-07 00:17:12 +00:00
|
|
|
chan->stream = NULL;
|
|
|
|
}
|
|
|
|
/* Close video stream */
|
|
|
|
if (chan->vstream) {
|
2004-04-02 23:25:24 +00:00
|
|
|
ast_closestream(chan->vstream);
|
2006-08-07 00:17:12 +00:00
|
|
|
chan->vstream = NULL;
|
|
|
|
}
|
Merged revisions 7265-7266,7268-7275 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r7265 | oej | 2005-12-01 17:18:14 -0600 (Thu, 01 Dec 2005) | 2 lines
Changing bug report address to the Asterisk issue tracker
........
r7266 | kpfleming | 2005-12-01 17:18:29 -0600 (Thu, 01 Dec 2005) | 3 lines
Makefile 'update' target now supports updating from Subversion repositories (issue #5875)
remove support for 'patches' subdirectory, it's no longer useful
........
r7268 | kpfleming | 2005-12-01 17:34:58 -0600 (Thu, 01 Dec 2005) | 2 lines
ensure channel's scheduling context is freed (issue #5788)
........
r7269 | kpfleming | 2005-12-01 17:49:44 -0600 (Thu, 01 Dec 2005) | 2 lines
don't block waiting for the Festival server forever when it goes away (issue #5882)
........
r7270 | kpfleming | 2005-12-01 18:26:12 -0600 (Thu, 01 Dec 2005) | 2 lines
allow variables to exist on both 'halves' of the Local channel (issue #5810)
........
r7271 | kpfleming | 2005-12-01 18:28:48 -0600 (Thu, 01 Dec 2005) | 2 lines
protect agent_bridgedchannel() from segfaulting when there is no bridged channel (issue #5879)
........
r7272 | kpfleming | 2005-12-01 18:39:00 -0600 (Thu, 01 Dec 2005) | 3 lines
properly handle password changes when mailbox is last line of config file and not followed by a newline (issue #5870)
reformat password changing code to conform to coding guidelines (issue #5870)
........
r7273 | kpfleming | 2005-12-01 18:42:40 -0600 (Thu, 01 Dec 2005) | 2 lines
allow previous context-searching behavior to be used if desired (issue #5899)
........
r7274 | kpfleming | 2005-12-01 18:51:15 -0600 (Thu, 01 Dec 2005) | 2 lines
inherit channel variables into channels created by Page() application (issue #5888)
........
r7275 | oej | 2005-12-01 18:52:13 -0600 (Thu, 01 Dec 2005) | 2 lines
Bug #5907. Improve SIP INFO DTMF debugging output. (1.2 & Trunk)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7276 f38db490-d61c-443f-a65b-d21fe96a405b
2005-12-02 01:01:11 +00:00
|
|
|
if (chan->sched) {
|
1999-12-04 21:35:07 +00:00
|
|
|
sched_context_destroy(chan->sched);
|
Merged revisions 7265-7266,7268-7275 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r7265 | oej | 2005-12-01 17:18:14 -0600 (Thu, 01 Dec 2005) | 2 lines
Changing bug report address to the Asterisk issue tracker
........
r7266 | kpfleming | 2005-12-01 17:18:29 -0600 (Thu, 01 Dec 2005) | 3 lines
Makefile 'update' target now supports updating from Subversion repositories (issue #5875)
remove support for 'patches' subdirectory, it's no longer useful
........
r7268 | kpfleming | 2005-12-01 17:34:58 -0600 (Thu, 01 Dec 2005) | 2 lines
ensure channel's scheduling context is freed (issue #5788)
........
r7269 | kpfleming | 2005-12-01 17:49:44 -0600 (Thu, 01 Dec 2005) | 2 lines
don't block waiting for the Festival server forever when it goes away (issue #5882)
........
r7270 | kpfleming | 2005-12-01 18:26:12 -0600 (Thu, 01 Dec 2005) | 2 lines
allow variables to exist on both 'halves' of the Local channel (issue #5810)
........
r7271 | kpfleming | 2005-12-01 18:28:48 -0600 (Thu, 01 Dec 2005) | 2 lines
protect agent_bridgedchannel() from segfaulting when there is no bridged channel (issue #5879)
........
r7272 | kpfleming | 2005-12-01 18:39:00 -0600 (Thu, 01 Dec 2005) | 3 lines
properly handle password changes when mailbox is last line of config file and not followed by a newline (issue #5870)
reformat password changing code to conform to coding guidelines (issue #5870)
........
r7273 | kpfleming | 2005-12-01 18:42:40 -0600 (Thu, 01 Dec 2005) | 2 lines
allow previous context-searching behavior to be used if desired (issue #5899)
........
r7274 | kpfleming | 2005-12-01 18:51:15 -0600 (Thu, 01 Dec 2005) | 2 lines
inherit channel variables into channels created by Page() application (issue #5888)
........
r7275 | oej | 2005-12-01 18:52:13 -0600 (Thu, 01 Dec 2005) | 2 lines
Bug #5907. Improve SIP INFO DTMF debugging output. (1.2 & Trunk)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7276 f38db490-d61c-443f-a65b-d21fe96a405b
2005-12-02 01:01:11 +00:00
|
|
|
chan->sched = NULL;
|
|
|
|
}
|
2005-05-15 04:48:30 +00:00
|
|
|
|
2006-04-06 15:40:12 +00:00
|
|
|
if (chan->generatordata) /* Clear any tone stuff remaining */
|
2006-12-21 19:36:42 +00:00
|
|
|
if (chan->generator && chan->generator->release)
|
|
|
|
chan->generator->release(chan, chan->generatordata);
|
2002-06-21 01:40:13 +00:00
|
|
|
chan->generatordata = NULL;
|
|
|
|
chan->generator = NULL;
|
2004-12-07 20:38:43 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_BLOCKING)) {
|
2001-03-12 03:04:51 +00:00
|
|
|
ast_log(LOG_WARNING, "Hard hangup called by thread %ld on %s, while fd "
|
2004-04-27 21:21:57 +00:00
|
|
|
"is blocked by thread %ld in procedure %s! Expect a failure\n",
|
2003-09-08 16:48:07 +00:00
|
|
|
(long)pthread_self(), chan->name, (long)chan->blocker, chan->blockproc);
|
2008-06-11 18:19:24 +00:00
|
|
|
ast_assert(ast_test_flag(chan, AST_FLAG_BLOCKING) == 0);
|
2001-03-12 03:04:51 +00:00
|
|
|
}
|
2004-12-07 20:38:43 +00:00
|
|
|
if (!ast_test_flag(chan, AST_FLAG_ZOMBIE)) {
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Hanging up channel '%s'\n", chan->name);
|
2005-03-04 06:47:24 +00:00
|
|
|
if (chan->tech->hangup)
|
|
|
|
res = chan->tech->hangup(chan);
|
2005-07-25 22:13:32 +00:00
|
|
|
} else {
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Hanging up zombie '%s'\n", chan->name);
|
2005-07-25 22:13:32 +00:00
|
|
|
}
|
2001-05-07 00:43:32 +00:00
|
|
|
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_unlock(chan);
|
2006-04-06 15:40:12 +00:00
|
|
|
manager_event(EVENT_FLAG_CALL, "Hangup",
|
2003-06-11 12:24:13 +00:00
|
|
|
"Channel: %s\r\n"
|
2004-06-22 20:11:15 +00:00
|
|
|
"Uniqueid: %s\r\n"
|
2007-11-14 22:59:05 +00:00
|
|
|
"CallerIDNum: %s\r\n"
|
|
|
|
"CallerIDName: %s\r\n"
|
2005-07-25 22:13:32 +00:00
|
|
|
"Cause: %d\r\n"
|
|
|
|
"Cause-txt: %s\r\n",
|
2006-04-06 15:40:12 +00:00
|
|
|
chan->name,
|
|
|
|
chan->uniqueid,
|
2007-11-14 23:08:54 +00:00
|
|
|
S_OR(chan->cid.cid_num, "<unknown>"),
|
|
|
|
S_OR(chan->cid.cid_name, "<unknown>"),
|
2005-07-25 22:13:32 +00:00
|
|
|
chan->hangupcause,
|
|
|
|
ast_cause2str(chan->hangupcause)
|
|
|
|
);
|
2007-06-27 20:26:53 +00:00
|
|
|
|
Merged revisions 135799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines
(closes issue #12982)
Reported by: bcnit
Tested by: murf
I discovered that also, in the previous bug fixes and changes,
the cdr.conf 'unanswered' option is not being obeyed, so
I fixed this.
And, yes, there are two 'answer' times involved in this
scenario, and I would agree with you, that the first
answer time is the time that should appear in the CDR.
(the second 'answer' time is the time that the bridge
was begun).
I made the necessary adjustments, recording the first
answer time into the peer cdr, and then using that to
override the bridge cdr's value.
To get the 'unanswered' CDRs to appear, I purposely
output them, using the dial cmd to mark them as
DIALED (with a new flag), and outputting them if
they bear that flag, and you are in the right mode.
I also corrected one small mention of the Zap device
to equally consider the dahdi device.
I heavily tested 10-sec-wait macros in dial, and
without the macro call; I tested hangups while the
macro was running vs. letting the macro complete
and the bridge form. Looks OK. Removed all the
instrumentation and debug.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135821 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05 23:45:32 +00:00
|
|
|
if (chan->cdr && !ast_test_flag(chan->cdr, AST_CDR_FLAG_BRIDGED) &&
|
|
|
|
!ast_test_flag(chan->cdr, AST_CDR_FLAG_POST_DISABLED) &&
|
|
|
|
(chan->cdr->disposition != AST_CDR_NULL || ast_test_flag(chan->cdr, AST_CDR_FLAG_DIALED))) {
|
|
|
|
|
2008-07-03 17:16:44 +00:00
|
|
|
ast_cdr_end(chan->cdr);
|
|
|
|
ast_cdr_detach(chan->cdr);
|
2008-09-02 22:50:59 +00:00
|
|
|
chan->cdr = NULL;
|
2008-07-03 17:16:44 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
ast_channel_free(chan);
|
2007-06-27 20:26:53 +00:00
|
|
|
|
1999-12-04 21:35:07 +00:00
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
2008-08-07 19:36:46 +00:00
|
|
|
#define ANSWER_WAIT_MS 500
|
2007-01-12 15:01:46 +00:00
|
|
|
int __ast_answer(struct ast_channel *chan, unsigned int delay)
|
1999-12-04 21:35:07 +00:00
|
|
|
{
|
2001-09-19 14:40:16 +00:00
|
|
|
int res = 0;
|
2007-01-11 23:42:14 +00:00
|
|
|
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_lock(chan);
|
2007-01-11 23:42:14 +00:00
|
|
|
|
2006-05-09 15:01:10 +00:00
|
|
|
/* You can't answer an outbound call */
|
2006-06-06 05:27:14 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_OUTGOING)) {
|
|
|
|
ast_channel_unlock(chan);
|
2006-05-09 15:01:10 +00:00
|
|
|
return 0;
|
2006-06-06 05:27:14 +00:00
|
|
|
}
|
2007-01-11 23:42:14 +00:00
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Stop if we're a zombie or need a soft hangup */
|
2004-12-07 20:38:43 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_ZOMBIE) || ast_check_hangup(chan)) {
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_unlock(chan);
|
2001-05-07 00:43:32 +00:00
|
|
|
return -1;
|
2004-06-23 17:41:51 +00:00
|
|
|
}
|
2007-01-11 23:42:14 +00:00
|
|
|
|
2008-09-09 01:47:56 +00:00
|
|
|
ast_channel_unlock(chan);
|
|
|
|
|
2007-01-23 00:11:32 +00:00
|
|
|
switch (chan->_state) {
|
2001-05-07 00:43:32 +00:00
|
|
|
case AST_STATE_RINGING:
|
2001-09-19 14:40:16 +00:00
|
|
|
case AST_STATE_RING:
|
2008-09-09 01:47:56 +00:00
|
|
|
ast_channel_lock(chan);
|
|
|
|
if (chan->tech->answer) {
|
2008-08-07 19:36:46 +00:00
|
|
|
res = chan->tech->answer(chan);
|
2008-09-09 01:47:56 +00:00
|
|
|
}
|
2008-08-07 19:36:46 +00:00
|
|
|
ast_setstate(chan, AST_STATE_UP);
|
|
|
|
ast_cdr_answer(chan->cdr);
|
2008-09-09 01:47:56 +00:00
|
|
|
ast_channel_unlock(chan);
|
|
|
|
if (delay) {
|
2007-01-12 15:01:46 +00:00
|
|
|
ast_safe_sleep(chan, delay);
|
2008-09-09 01:47:56 +00:00
|
|
|
} else {
|
2008-08-07 19:36:46 +00:00
|
|
|
struct ast_frame *f;
|
2008-08-07 19:54:27 +00:00
|
|
|
int ms = ANSWER_WAIT_MS;
|
2008-08-07 19:36:46 +00:00
|
|
|
while (1) {
|
|
|
|
/* 500 ms was the original delay here, so now
|
|
|
|
* we cap our waiting at 500 ms
|
|
|
|
*/
|
2008-08-07 19:54:27 +00:00
|
|
|
ms = ast_waitfor(chan, ms);
|
|
|
|
if (ms < 0) {
|
2008-08-07 19:36:46 +00:00
|
|
|
ast_log(LOG_WARNING, "Error condition occurred when polling channel %s for a voice frame: %s\n", chan->name, strerror(errno));
|
2008-08-07 19:54:27 +00:00
|
|
|
res = -1;
|
2008-08-07 19:36:46 +00:00
|
|
|
break;
|
|
|
|
}
|
2008-08-07 19:54:27 +00:00
|
|
|
if (ms == 0) {
|
2008-08-07 19:36:46 +00:00
|
|
|
ast_debug(2, "Didn't receive a voice frame from %s within %d ms of answering. Continuing anyway\n", chan->name, ANSWER_WAIT_MS);
|
2008-08-07 19:54:27 +00:00
|
|
|
res = 0;
|
2008-08-07 19:36:46 +00:00
|
|
|
break;
|
|
|
|
}
|
|
|
|
f = ast_read(chan);
|
|
|
|
if (!f || (f->frametype == AST_FRAME_CONTROL && f->subclass == AST_CONTROL_HANGUP)) {
|
|
|
|
res = -1;
|
|
|
|
ast_debug(2, "Hangup of channel %s detected in answer routine\n", chan->name);
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
if (f->frametype == AST_FRAME_VOICE) {
|
|
|
|
res = 0;
|
|
|
|
break;
|
|
|
|
}
|
2008-07-01 19:53:03 +00:00
|
|
|
}
|
|
|
|
}
|
2008-08-07 19:36:46 +00:00
|
|
|
break;
|
2001-05-07 00:43:32 +00:00
|
|
|
case AST_STATE_UP:
|
|
|
|
break;
|
2006-08-19 03:50:15 +00:00
|
|
|
default:
|
|
|
|
break;
|
1999-12-04 21:35:07 +00:00
|
|
|
}
|
2008-09-09 01:47:56 +00:00
|
|
|
|
2007-12-03 18:44:16 +00:00
|
|
|
chan->visible_indication = 0;
|
2007-01-11 23:42:14 +00:00
|
|
|
|
2006-04-16 15:22:13 +00:00
|
|
|
return res;
|
1999-12-04 21:35:07 +00:00
|
|
|
}
|
|
|
|
|
2007-01-12 15:01:46 +00:00
|
|
|
int ast_answer(struct ast_channel *chan)
|
|
|
|
{
|
2008-08-07 19:36:46 +00:00
|
|
|
return __ast_answer(chan, 0);
|
2007-01-12 15:01:46 +00:00
|
|
|
}
|
|
|
|
|
2004-04-07 14:43:18 +00:00
|
|
|
void ast_deactivate_generator(struct ast_channel *chan)
|
2002-06-21 01:40:13 +00:00
|
|
|
{
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_lock(chan);
|
2002-06-21 01:40:13 +00:00
|
|
|
if (chan->generatordata) {
|
2006-04-06 15:40:12 +00:00
|
|
|
if (chan->generator && chan->generator->release)
|
2004-02-04 23:18:16 +00:00
|
|
|
chan->generator->release(chan, chan->generatordata);
|
2002-06-21 01:40:13 +00:00
|
|
|
chan->generatordata = NULL;
|
2002-09-10 20:48:20 +00:00
|
|
|
chan->generator = NULL;
|
2007-08-08 21:44:58 +00:00
|
|
|
ast_channel_set_fd(chan, AST_GENERATOR_FD, -1);
|
2004-12-07 20:38:43 +00:00
|
|
|
ast_clear_flag(chan, AST_FLAG_WRITE_INT);
|
2004-08-27 03:28:32 +00:00
|
|
|
ast_settimeout(chan, 0, NULL, NULL);
|
2002-06-21 01:40:13 +00:00
|
|
|
}
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_unlock(chan);
|
2004-04-02 21:58:10 +00:00
|
|
|
}
|
2004-04-07 14:43:18 +00:00
|
|
|
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2007-09-21 14:40:10 +00:00
|
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|
static int generator_force(const void *data)
|
2004-08-27 03:28:32 +00:00
|
|
|
{
|
|
|
|
/* Called if generator doesn't have data */
|
|
|
|
void *tmp;
|
|
|
|
int res;
|
2008-03-10 17:13:34 +00:00
|
|
|
int (*generate)(struct ast_channel *chan, void *tmp, int datalen, int samples) = NULL;
|
2007-09-21 14:40:10 +00:00
|
|
|
struct ast_channel *chan = (struct ast_channel *)data;
|
2008-01-28 17:21:24 +00:00
|
|
|
|
|
|
|
ast_channel_lock(chan);
|
2004-08-27 03:28:32 +00:00
|
|
|
tmp = chan->generatordata;
|
|
|
|
chan->generatordata = NULL;
|
2008-03-10 16:59:13 +00:00
|
|
|
if (chan->generator)
|
|
|
|
generate = chan->generator->generate;
|
2008-01-28 17:21:24 +00:00
|
|
|
ast_channel_unlock(chan);
|
|
|
|
|
2008-03-10 16:59:13 +00:00
|
|
|
if (!tmp || !generate)
|
|
|
|
return 0;
|
|
|
|
|
Add some fixes that I made in regards to wideband codec handling to get
G.722 music on hold working for me.
(issue #12164, reported by milazzo and jsmith, patches by me)
res/res_musiconhold.c:
- I moved a single line so that the sample queue update happened before
ast_write(). The reason that this was a bug is that the G.722 frame
originally says it has 320 samples in it (which is correct). However,
when the frame is written to a channel that uses RTP, main/rtp.c modifies
the frame to cut the number of samples in half before it sends it on
the wire. This is to account for the stupid incorrect G.722 spec that
makes it so we have to lie about the number of samples with RTP. I should
probably go and re-work the RTP code so it doesn't modify the frame so
that a bug like this won't happen in the future. However, this change to
MOH is harmless.
main/channel.c:
- I made two fixes in regards to generator timing. Generators use samples
for timing. However, this code assumed 8 kHz samples. In one case, it was
a hard coded 160 samples, that is now written as the sample rate / 50. The
other place was dealing with timing a generator based on frames coming from
the other direction. However, that would have only worked if the sample
rates for the formats in both directions were the same. The code now takes
into account that the sample rates may differ, and scales the generator
samples accordingly.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110268 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-20 17:41:22 +00:00
|
|
|
res = generate(chan, tmp, 0, ast_format_rate(chan->writeformat & AST_FORMAT_AUDIO_MASK) / 50);
|
2008-01-28 17:21:24 +00:00
|
|
|
|
2004-08-27 03:28:32 +00:00
|
|
|
chan->generatordata = tmp;
|
2008-01-28 17:21:24 +00:00
|
|
|
|
2004-08-27 03:28:32 +00:00
|
|
|
if (res) {
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Auto-deactivating generator\n");
|
2004-08-27 03:28:32 +00:00
|
|
|
ast_deactivate_generator(chan);
|
|
|
|
}
|
2008-01-28 17:21:24 +00:00
|
|
|
|
2004-08-27 03:28:32 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2002-06-21 01:40:13 +00:00
|
|
|
int ast_activate_generator(struct ast_channel *chan, struct ast_generator *gen, void *params)
|
|
|
|
{
|
2004-06-02 20:18:18 +00:00
|
|
|
int res = 0;
|
2005-10-28 23:01:13 +00:00
|
|
|
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_lock(chan);
|
2005-10-28 23:01:13 +00:00
|
|
|
|
2002-06-21 01:40:13 +00:00
|
|
|
if (chan->generatordata) {
|
2004-02-04 23:18:16 +00:00
|
|
|
if (chan->generator && chan->generator->release)
|
|
|
|
chan->generator->release(chan, chan->generatordata);
|
2002-06-21 01:40:13 +00:00
|
|
|
chan->generatordata = NULL;
|
|
|
|
}
|
2005-10-28 23:01:13 +00:00
|
|
|
|
2003-04-16 02:47:03 +00:00
|
|
|
ast_prod(chan);
|
2006-02-15 01:48:54 +00:00
|
|
|
if (gen->alloc && !(chan->generatordata = gen->alloc(chan, params))) {
|
|
|
|
res = -1;
|
2005-10-28 23:01:13 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
if (!res) {
|
2008-06-13 12:45:50 +00:00
|
|
|
ast_settimeout(chan, 50, generator_force, chan);
|
2002-06-21 01:40:13 +00:00
|
|
|
chan->generator = gen;
|
|
|
|
}
|
2005-10-28 23:01:13 +00:00
|
|
|
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_unlock(chan);
|
2005-10-28 23:01:13 +00:00
|
|
|
|
2004-06-02 20:18:18 +00:00
|
|
|
return res;
|
2002-06-21 01:40:13 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Wait for x amount of time on a file descriptor to have input. */
|
2001-03-12 03:04:51 +00:00
|
|
|
int ast_waitfor_n_fd(int *fds, int n, int *ms, int *exception)
|
1999-12-04 21:35:07 +00:00
|
|
|
{
|
|
|
|
int winner = -1;
|
2006-01-30 03:13:33 +00:00
|
|
|
ast_waitfor_nandfds(NULL, 0, fds, n, exception, &winner, ms);
|
1999-12-04 21:35:07 +00:00
|
|
|
return winner;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Wait for x amount of time on a file descriptor to have input. */
|
2007-08-08 21:44:58 +00:00
|
|
|
#ifdef HAVE_EPOLL
|
|
|
|
static struct ast_channel *ast_waitfor_nandfds_classic(struct ast_channel **c, int n, int *fds, int nfds,
|
|
|
|
int *exception, int *outfd, int *ms)
|
|
|
|
#else
|
2006-04-06 15:40:12 +00:00
|
|
|
struct ast_channel *ast_waitfor_nandfds(struct ast_channel **c, int n, int *fds, int nfds,
|
2007-08-08 21:44:58 +00:00
|
|
|
int *exception, int *outfd, int *ms)
|
|
|
|
#endif
|
1999-12-04 21:35:07 +00:00
|
|
|
{
|
2005-07-27 22:19:49 +00:00
|
|
|
struct timeval start = { 0 , 0 };
|
3) In addition to merging the changes below, change trunk back to a regular
LIST instead of an RWLIST. The way this list works makes it such that
a RWLIST provides no additional benefit. Also, a mutex is needed for
use with the thread condition.
Merged revisions 105563 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) | 24 lines
Merge in some changes from team/russell/autoservice-nochans-1.4
These changes fix up some dubious code that I came across while auditing what
happens in the autoservice thread when there are no channels currently in
autoservice.
1) Change it so that autoservice thread doesn't keep looping around calling
ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition
so that the thread properly goes to sleep and does not wake up until a
channel is put into autoservice.
This actually fixes an interesting bug, as well. If the autoservice thread
is already running (almost always is the case), then when the thread goes
from having 0 channels to have 1 channel to autoservice, that channel would
have to wait for up to 1/2 of a second to have the first frame read from it.
2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no
channels and no fds to poll() on, such as was the case with the previous code
for the autoservice thread. In this case, the code would call alloca(0), and
pass the result as the first argument to poll(). In this case, the 2nd
argument to poll() specified that there were no fds, so this invalid pointer
shouldn't actually get dereferenced, but, this code makes it explicit and
ensures the pointers are NULL unless we have valid data to put there.
(related to issue #12116)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105564 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-03 15:59:50 +00:00
|
|
|
struct pollfd *pfds = NULL;
|
1999-12-04 21:35:07 +00:00
|
|
|
int res;
|
2004-04-25 20:42:45 +00:00
|
|
|
long rms;
|
|
|
|
int x, y, max;
|
2006-01-30 03:13:33 +00:00
|
|
|
int sz;
|
2008-05-01 23:06:23 +00:00
|
|
|
struct timeval now = { 0, 0 };
|
|
|
|
struct timeval whentohangup = { 0, 0 }, diff;
|
1999-12-04 21:35:07 +00:00
|
|
|
struct ast_channel *winner = NULL;
|
2006-01-30 03:13:33 +00:00
|
|
|
struct fdmap {
|
|
|
|
int chan;
|
|
|
|
int fdno;
|
3) In addition to merging the changes below, change trunk back to a regular
LIST instead of an RWLIST. The way this list works makes it such that
a RWLIST provides no additional benefit. Also, a mutex is needed for
use with the thread condition.
Merged revisions 105563 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) | 24 lines
Merge in some changes from team/russell/autoservice-nochans-1.4
These changes fix up some dubious code that I came across while auditing what
happens in the autoservice thread when there are no channels currently in
autoservice.
1) Change it so that autoservice thread doesn't keep looping around calling
ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition
so that the thread properly goes to sleep and does not wake up until a
channel is put into autoservice.
This actually fixes an interesting bug, as well. If the autoservice thread
is already running (almost always is the case), then when the thread goes
from having 0 channels to have 1 channel to autoservice, that channel would
have to wait for up to 1/2 of a second to have the first frame read from it.
2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no
channels and no fds to poll() on, such as was the case with the previous code
for the autoservice thread. In this case, the code would call alloca(0), and
pass the result as the first argument to poll(). In this case, the 2nd
argument to poll() specified that there were no fds, so this invalid pointer
shouldn't actually get dereferenced, but, this code makes it explicit and
ensures the pointers are NULL unless we have valid data to put there.
(related to issue #12116)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105564 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-03 15:59:50 +00:00
|
|
|
} *fdmap = NULL;
|
2006-01-30 03:13:33 +00:00
|
|
|
|
3) In addition to merging the changes below, change trunk back to a regular
LIST instead of an RWLIST. The way this list works makes it such that
a RWLIST provides no additional benefit. Also, a mutex is needed for
use with the thread condition.
Merged revisions 105563 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) | 24 lines
Merge in some changes from team/russell/autoservice-nochans-1.4
These changes fix up some dubious code that I came across while auditing what
happens in the autoservice thread when there are no channels currently in
autoservice.
1) Change it so that autoservice thread doesn't keep looping around calling
ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition
so that the thread properly goes to sleep and does not wake up until a
channel is put into autoservice.
This actually fixes an interesting bug, as well. If the autoservice thread
is already running (almost always is the case), then when the thread goes
from having 0 channels to have 1 channel to autoservice, that channel would
have to wait for up to 1/2 of a second to have the first frame read from it.
2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no
channels and no fds to poll() on, such as was the case with the previous code
for the autoservice thread. In this case, the code would call alloca(0), and
pass the result as the first argument to poll(). In this case, the 2nd
argument to poll() specified that there were no fds, so this invalid pointer
shouldn't actually get dereferenced, but, this code makes it explicit and
ensures the pointers are NULL unless we have valid data to put there.
(related to issue #12116)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105564 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-03 15:59:50 +00:00
|
|
|
if ((sz = n * AST_MAX_FDS + nfds)) {
|
|
|
|
pfds = alloca(sizeof(*pfds) * sz);
|
|
|
|
fdmap = alloca(sizeof(*fdmap) * sz);
|
|
|
|
}
|
2004-04-25 20:42:45 +00:00
|
|
|
|
2001-09-19 14:40:16 +00:00
|
|
|
if (outfd)
|
2004-03-05 18:22:31 +00:00
|
|
|
*outfd = -99999;
|
2001-09-19 14:40:16 +00:00
|
|
|
if (exception)
|
|
|
|
*exception = 0;
|
1999-12-04 21:35:07 +00:00
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Perform any pending masquerades */
|
2007-08-08 21:44:58 +00:00
|
|
|
for (x = 0; x < n; x++) {
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_lock(c[x]);
|
2007-08-08 21:44:58 +00:00
|
|
|
if (c[x]->masq && ast_do_masquerade(c[x])) {
|
|
|
|
ast_log(LOG_WARNING, "Masquerade failed\n");
|
|
|
|
*ms = -1;
|
|
|
|
ast_channel_unlock(c[x]);
|
|
|
|
return NULL;
|
2001-05-07 00:43:32 +00:00
|
|
|
}
|
2008-05-01 23:06:23 +00:00
|
|
|
if (!ast_tvzero(c[x]->whentohangup)) {
|
|
|
|
if (ast_tvzero(whentohangup))
|
|
|
|
now = ast_tvnow();
|
|
|
|
diff = ast_tvsub(c[x]->whentohangup, now);
|
|
|
|
if (diff.tv_sec < 0 || ast_tvzero(diff)) {
|
2006-01-30 03:13:33 +00:00
|
|
|
/* Should already be hungup */
|
|
|
|
c[x]->_softhangup |= AST_SOFTHANGUP_TIMEOUT;
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_unlock(c[x]);
|
2006-01-30 03:13:33 +00:00
|
|
|
return c[x];
|
|
|
|
}
|
2008-05-01 23:06:23 +00:00
|
|
|
if (ast_tvzero(whentohangup) || ast_tvcmp(diff, whentohangup) < 0)
|
2006-01-30 03:13:33 +00:00
|
|
|
whentohangup = diff;
|
|
|
|
}
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_unlock(c[x]);
|
2001-05-07 00:43:32 +00:00
|
|
|
}
|
2006-01-30 03:13:33 +00:00
|
|
|
/* Wait full interval */
|
2004-04-25 20:42:45 +00:00
|
|
|
rms = *ms;
|
2008-05-01 23:06:23 +00:00
|
|
|
if (!ast_tvzero(whentohangup)) {
|
|
|
|
rms = whentohangup.tv_sec * 1000 + whentohangup.tv_usec / 1000; /* timeout in milliseconds */
|
2006-01-30 03:13:33 +00:00
|
|
|
if (*ms >= 0 && *ms < rms) /* original *ms still smaller */
|
|
|
|
rms = *ms;
|
2004-03-05 18:22:31 +00:00
|
|
|
}
|
2006-01-30 03:13:33 +00:00
|
|
|
/*
|
|
|
|
* Build the pollfd array, putting the channels' fds first,
|
|
|
|
* followed by individual fds. Order is important because
|
|
|
|
* individual fd's must have priority over channel fds.
|
|
|
|
*/
|
2004-04-25 20:42:45 +00:00
|
|
|
max = 0;
|
2007-08-08 21:44:58 +00:00
|
|
|
for (x = 0; x < n; x++) {
|
|
|
|
for (y = 0; y < AST_MAX_FDS; y++) {
|
2006-01-30 03:13:33 +00:00
|
|
|
fdmap[max].fdno = y; /* fd y is linked to this pfds */
|
|
|
|
fdmap[max].chan = x; /* channel x is linked to this pfds */
|
|
|
|
max += ast_add_fd(&pfds[max], c[x]->fds[y]);
|
2001-05-07 00:43:32 +00:00
|
|
|
}
|
1999-12-04 21:35:07 +00:00
|
|
|
CHECK_BLOCKING(c[x]);
|
|
|
|
}
|
2006-01-30 03:13:33 +00:00
|
|
|
/* Add the individual fds */
|
2007-08-08 21:44:58 +00:00
|
|
|
for (x = 0; x < nfds; x++) {
|
2006-01-30 03:13:33 +00:00
|
|
|
fdmap[max].chan = -1;
|
|
|
|
max += ast_add_fd(&pfds[max], fds[x]);
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
2006-01-30 03:13:33 +00:00
|
|
|
|
2006-04-06 15:40:12 +00:00
|
|
|
if (*ms > 0)
|
2005-07-15 23:00:47 +00:00
|
|
|
start = ast_tvnow();
|
2005-10-20 22:45:47 +00:00
|
|
|
|
2006-01-30 03:13:33 +00:00
|
|
|
if (sizeof(int) == 4) { /* XXX fix timeout > 600000 on linux x86-32 */
|
2005-10-20 22:45:47 +00:00
|
|
|
do {
|
|
|
|
int kbrms = rms;
|
|
|
|
if (kbrms > 600000)
|
|
|
|
kbrms = 600000;
|
|
|
|
res = poll(pfds, max, kbrms);
|
|
|
|
if (!res)
|
|
|
|
rms -= kbrms;
|
|
|
|
} while (!res && (rms > 0));
|
|
|
|
} else {
|
|
|
|
res = poll(pfds, max, rms);
|
|
|
|
}
|
2007-08-08 21:44:58 +00:00
|
|
|
for (x = 0; x < n; x++)
|
2006-01-30 03:13:33 +00:00
|
|
|
ast_clear_flag(c[x], AST_FLAG_BLOCKING);
|
|
|
|
if (res < 0) { /* Simulate a timeout if we were interrupted */
|
2006-04-06 08:02:35 +00:00
|
|
|
if (errno != EINTR)
|
|
|
|
*ms = -1;
|
2001-05-07 00:43:32 +00:00
|
|
|
return NULL;
|
|
|
|
}
|
2008-05-01 23:06:23 +00:00
|
|
|
if (!ast_tvzero(whentohangup)) { /* if we have a timeout, check who expired */
|
|
|
|
now = ast_tvnow();
|
2007-08-08 21:44:58 +00:00
|
|
|
for (x = 0; x < n; x++) {
|
2008-05-01 23:06:23 +00:00
|
|
|
if (!ast_tvzero(c[x]->whentohangup) && ast_tvcmp(c[x]->whentohangup, now) <= 0) {
|
2006-01-30 03:13:33 +00:00
|
|
|
c[x]->_softhangup |= AST_SOFTHANGUP_TIMEOUT;
|
|
|
|
if (winner == NULL)
|
2001-05-07 00:43:32 +00:00
|
|
|
winner = c[x];
|
|
|
|
}
|
2001-03-12 03:04:51 +00:00
|
|
|
}
|
1999-12-04 21:35:07 +00:00
|
|
|
}
|
2006-01-30 03:13:33 +00:00
|
|
|
if (res == 0) { /* no fd ready, reset timeout and done */
|
|
|
|
*ms = 0; /* XXX use 0 since we may not have an exact timeout. */
|
|
|
|
return winner;
|
|
|
|
}
|
|
|
|
/*
|
|
|
|
* Then check if any channel or fd has a pending event.
|
|
|
|
* Remember to check channels first and fds last, as they
|
|
|
|
* must have priority on setting 'winner'
|
|
|
|
*/
|
|
|
|
for (x = 0; x < max; x++) {
|
|
|
|
res = pfds[x].revents;
|
|
|
|
if (res == 0)
|
|
|
|
continue;
|
|
|
|
if (fdmap[x].chan >= 0) { /* this is a channel */
|
|
|
|
winner = c[fdmap[x].chan]; /* override previous winners */
|
|
|
|
if (res & POLLPRI)
|
|
|
|
ast_set_flag(winner, AST_FLAG_EXCEPTION);
|
|
|
|
else
|
|
|
|
ast_clear_flag(winner, AST_FLAG_EXCEPTION);
|
|
|
|
winner->fdno = fdmap[x].fdno;
|
|
|
|
} else { /* this is an fd */
|
|
|
|
if (outfd)
|
|
|
|
*outfd = pfds[x].fd;
|
|
|
|
if (exception)
|
|
|
|
*exception = (res & POLLPRI) ? -1 : 0;
|
|
|
|
winner = NULL;
|
|
|
|
}
|
2004-04-25 20:42:45 +00:00
|
|
|
}
|
|
|
|
if (*ms > 0) {
|
2005-07-15 23:00:47 +00:00
|
|
|
*ms -= ast_tvdiff_ms(ast_tvnow(), start);
|
|
|
|
if (*ms < 0)
|
2004-04-25 20:42:45 +00:00
|
|
|
*ms = 0;
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
1999-12-04 21:35:07 +00:00
|
|
|
return winner;
|
|
|
|
}
|
|
|
|
|
2007-08-08 21:44:58 +00:00
|
|
|
#ifdef HAVE_EPOLL
|
|
|
|
static struct ast_channel *ast_waitfor_nandfds_simple(struct ast_channel *chan, int *ms)
|
|
|
|
{
|
|
|
|
struct timeval start = { 0 , 0 };
|
|
|
|
int res = 0;
|
|
|
|
struct epoll_event ev[1];
|
2008-05-01 23:06:23 +00:00
|
|
|
long diff, rms = *ms;
|
2007-08-08 21:44:58 +00:00
|
|
|
struct ast_channel *winner = NULL;
|
|
|
|
struct ast_epoll_data *aed = NULL;
|
|
|
|
|
|
|
|
ast_channel_lock(chan);
|
|
|
|
|
|
|
|
/* See if this channel needs to be masqueraded */
|
|
|
|
if (chan->masq && ast_do_masquerade(chan)) {
|
|
|
|
ast_log(LOG_WARNING, "Failed to perform masquerade on %s\n", chan->name);
|
|
|
|
*ms = -1;
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Figure out their timeout */
|
2008-05-01 23:06:23 +00:00
|
|
|
if (!ast_tvzero(chan->whentohangup)) {
|
|
|
|
if ((diff = ast_tvdiff_ms(chan->whentohangup, ast_tvnow())) < 0) {
|
2007-08-08 21:44:58 +00:00
|
|
|
/* They should already be hungup! */
|
|
|
|
chan->_softhangup |= AST_SOFTHANGUP_TIMEOUT;
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
/* If this value is smaller then the current one... make it priority */
|
2008-05-01 23:06:23 +00:00
|
|
|
if (rms > diff)
|
|
|
|
rms = diff;
|
2007-08-08 21:44:58 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
|
|
|
|
/* Time to make this channel block... */
|
|
|
|
CHECK_BLOCKING(chan);
|
|
|
|
|
|
|
|
if (*ms > 0)
|
|
|
|
start = ast_tvnow();
|
|
|
|
|
|
|
|
/* We don't have to add any file descriptors... they are already added, we just have to wait! */
|
|
|
|
res = epoll_wait(chan->epfd, ev, 1, rms);
|
|
|
|
|
|
|
|
/* Stop blocking */
|
|
|
|
ast_clear_flag(chan, AST_FLAG_BLOCKING);
|
|
|
|
|
|
|
|
/* Simulate a timeout if we were interrupted */
|
|
|
|
if (res < 0) {
|
|
|
|
if (errno != EINTR)
|
|
|
|
*ms = -1;
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* If this channel has a timeout see if it expired */
|
2008-05-01 23:06:23 +00:00
|
|
|
if (!ast_tvzero(chan->whentohangup)) {
|
|
|
|
if (ast_tvdiff_ms(ast_tvnow(), chan->whentohangup) >= 0) {
|
2007-08-08 21:44:58 +00:00
|
|
|
chan->_softhangup |= AST_SOFTHANGUP_TIMEOUT;
|
|
|
|
winner = chan;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* No fd ready, reset timeout and be done for now */
|
|
|
|
if (!res) {
|
|
|
|
*ms = 0;
|
|
|
|
return winner;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* See what events are pending */
|
|
|
|
aed = ev[0].data.ptr;
|
|
|
|
chan->fdno = aed->which;
|
|
|
|
if (ev[0].events & EPOLLPRI)
|
|
|
|
ast_set_flag(chan, AST_FLAG_EXCEPTION);
|
|
|
|
else
|
|
|
|
ast_clear_flag(chan, AST_FLAG_EXCEPTION);
|
|
|
|
|
|
|
|
if (*ms > 0) {
|
|
|
|
*ms -= ast_tvdiff_ms(ast_tvnow(), start);
|
|
|
|
if (*ms < 0)
|
|
|
|
*ms = 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
return chan;
|
|
|
|
}
|
|
|
|
|
|
|
|
static struct ast_channel *ast_waitfor_nandfds_complex(struct ast_channel **c, int n, int *ms)
|
|
|
|
{
|
|
|
|
struct timeval start = { 0 , 0 };
|
|
|
|
int res = 0, i;
|
|
|
|
struct epoll_event ev[25] = { { 0, } };
|
2008-05-01 23:06:23 +00:00
|
|
|
struct timeval now = { 0, 0 };
|
|
|
|
long whentohangup = 0, diff = 0, rms = *ms;
|
2007-08-08 21:44:58 +00:00
|
|
|
struct ast_channel *winner = NULL;
|
|
|
|
|
|
|
|
for (i = 0; i < n; i++) {
|
|
|
|
ast_channel_lock(c[i]);
|
|
|
|
if (c[i]->masq && ast_do_masquerade(c[i])) {
|
|
|
|
ast_log(LOG_WARNING, "Masquerade failed\n");
|
|
|
|
*ms = -1;
|
|
|
|
ast_channel_unlock(c[i]);
|
|
|
|
return NULL;
|
|
|
|
}
|
2008-05-01 23:06:23 +00:00
|
|
|
if (!ast_tvzero(c[i]->whentohangup)) {
|
|
|
|
if (whentohangup == 0)
|
|
|
|
now = ast_tvnow();
|
|
|
|
if ((diff = ast_tvdiff_ms(c[i]->whentohangup, now)) < 0) {
|
2007-08-08 21:44:58 +00:00
|
|
|
c[i]->_softhangup |= AST_SOFTHANGUP_TIMEOUT;
|
|
|
|
ast_channel_unlock(c[i]);
|
|
|
|
return c[i];
|
|
|
|
}
|
2008-05-01 23:06:23 +00:00
|
|
|
if (!whentohangup || whentohangup > diff)
|
2007-08-08 21:44:58 +00:00
|
|
|
whentohangup = diff;
|
|
|
|
}
|
|
|
|
ast_channel_unlock(c[i]);
|
|
|
|
CHECK_BLOCKING(c[i]);
|
|
|
|
}
|
|
|
|
|
|
|
|
rms = *ms;
|
|
|
|
if (whentohangup) {
|
2008-05-01 23:06:23 +00:00
|
|
|
rms = whentohangup;
|
2007-08-08 21:44:58 +00:00
|
|
|
if (*ms >= 0 && *ms < rms)
|
|
|
|
rms = *ms;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (*ms > 0)
|
|
|
|
start = ast_tvnow();
|
|
|
|
|
|
|
|
res = epoll_wait(c[0]->epfd, ev, 25, rms);
|
|
|
|
|
|
|
|
for (i = 0; i < n; i++)
|
|
|
|
ast_clear_flag(c[i], AST_FLAG_BLOCKING);
|
|
|
|
|
|
|
|
if (res < 0) {
|
|
|
|
if (errno != EINTR)
|
|
|
|
*ms = -1;
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (whentohangup) {
|
2008-05-01 23:06:23 +00:00
|
|
|
now = ast_tvnow();
|
2007-08-08 21:44:58 +00:00
|
|
|
for (i = 0; i < n; i++) {
|
2008-05-01 23:06:23 +00:00
|
|
|
if (!ast_tvzero(c[i]->whentohangup) && ast_tvdiff_ms(now, c[i]->whentohangup) >= 0) {
|
2007-08-08 21:44:58 +00:00
|
|
|
c[i]->_softhangup |= AST_SOFTHANGUP_TIMEOUT;
|
|
|
|
if (!winner)
|
|
|
|
winner = c[i];
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
if (!res) {
|
|
|
|
*ms = 0;
|
|
|
|
return winner;
|
|
|
|
}
|
|
|
|
|
2007-12-06 04:37:36 +00:00
|
|
|
for (i = 0; i < res; i++) {
|
2007-08-08 21:44:58 +00:00
|
|
|
struct ast_epoll_data *aed = ev[i].data.ptr;
|
|
|
|
|
|
|
|
if (!ev[i].events || !aed)
|
|
|
|
continue;
|
|
|
|
|
|
|
|
winner = aed->chan;
|
|
|
|
if (ev[i].events & EPOLLPRI)
|
|
|
|
ast_set_flag(winner, AST_FLAG_EXCEPTION);
|
|
|
|
else
|
|
|
|
ast_clear_flag(winner, AST_FLAG_EXCEPTION);
|
|
|
|
winner->fdno = aed->which;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (*ms > 0) {
|
|
|
|
*ms -= ast_tvdiff_ms(ast_tvnow(), start);
|
|
|
|
if (*ms < 0)
|
|
|
|
*ms = 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
return winner;
|
|
|
|
}
|
|
|
|
|
|
|
|
struct ast_channel *ast_waitfor_nandfds(struct ast_channel **c, int n, int *fds, int nfds,
|
|
|
|
int *exception, int *outfd, int *ms)
|
|
|
|
{
|
2007-08-14 14:17:43 +00:00
|
|
|
/* Clear all provided values in one place. */
|
|
|
|
if (outfd)
|
|
|
|
*outfd = -99999;
|
|
|
|
if (exception)
|
|
|
|
*exception = 0;
|
|
|
|
|
2007-08-08 21:44:58 +00:00
|
|
|
/* If no epoll file descriptor is available resort to classic nandfds */
|
|
|
|
if (!n || nfds || c[0]->epfd == -1)
|
|
|
|
return ast_waitfor_nandfds_classic(c, n, fds, nfds, exception, outfd, ms);
|
|
|
|
else if (!nfds && n == 1)
|
|
|
|
return ast_waitfor_nandfds_simple(c[0], ms);
|
|
|
|
else
|
|
|
|
return ast_waitfor_nandfds_complex(c, n, ms);
|
|
|
|
}
|
|
|
|
#endif
|
|
|
|
|
2001-09-19 14:40:16 +00:00
|
|
|
struct ast_channel *ast_waitfor_n(struct ast_channel **c, int n, int *ms)
|
|
|
|
{
|
|
|
|
return ast_waitfor_nandfds(c, n, NULL, 0, NULL, NULL, ms);
|
|
|
|
}
|
|
|
|
|
1999-12-04 21:35:07 +00:00
|
|
|
int ast_waitfor(struct ast_channel *c, int ms)
|
|
|
|
{
|
2006-04-14 20:47:06 +00:00
|
|
|
int oldms = ms; /* -1 if no timeout */
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2006-01-30 03:13:33 +00:00
|
|
|
ast_waitfor_nandfds(&c, 1, NULL, 0, NULL, NULL, &ms);
|
|
|
|
if ((ms < 0) && (oldms < 0))
|
|
|
|
ms = 0;
|
2001-05-07 00:43:32 +00:00
|
|
|
return ms;
|
1999-12-04 21:35:07 +00:00
|
|
|
}
|
|
|
|
|
2006-04-14 20:47:06 +00:00
|
|
|
/* XXX never to be called with ms = -1 */
|
2005-06-05 15:04:43 +00:00
|
|
|
int ast_waitfordigit(struct ast_channel *c, int ms)
|
1999-12-04 21:35:07 +00:00
|
|
|
{
|
2006-04-16 07:53:25 +00:00
|
|
|
return ast_waitfordigit_full(c, ms, -1, -1);
|
1999-12-04 21:35:07 +00:00
|
|
|
}
|
|
|
|
|
2008-06-13 12:45:50 +00:00
|
|
|
int ast_settimeout(struct ast_channel *c, unsigned int rate, int (*func)(const void *data), void *data)
|
2003-05-12 04:23:55 +00:00
|
|
|
{
|
2008-06-13 12:45:50 +00:00
|
|
|
int res;
|
2008-06-26 15:37:01 +00:00
|
|
|
unsigned int real_rate = rate, max_rate;
|
2008-06-13 12:45:50 +00:00
|
|
|
|
|
|
|
if (c->timingfd == -1) {
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (!func) {
|
|
|
|
rate = 0;
|
|
|
|
data = NULL;
|
2003-05-12 04:23:55 +00:00
|
|
|
}
|
2008-06-13 12:45:50 +00:00
|
|
|
|
2008-06-26 15:37:01 +00:00
|
|
|
if (rate && rate > (max_rate = ast_timer_get_max_rate(c->timingfd))) {
|
|
|
|
real_rate = max_rate;
|
|
|
|
}
|
|
|
|
|
|
|
|
ast_debug(1, "Scheduling timer at (%u requested / %u actual) timer ticks per second\n", rate, real_rate);
|
2008-06-13 12:45:50 +00:00
|
|
|
|
2008-06-26 15:37:01 +00:00
|
|
|
res = ast_timer_set_rate(c->timingfd, real_rate);
|
2008-06-13 12:45:50 +00:00
|
|
|
|
|
|
|
c->timingfunc = func;
|
|
|
|
c->timingdata = data;
|
|
|
|
|
2003-05-12 04:23:55 +00:00
|
|
|
return res;
|
|
|
|
}
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2005-06-05 15:04:43 +00:00
|
|
|
int ast_waitfordigit_full(struct ast_channel *c, int ms, int audiofd, int cmdfd)
|
2003-02-23 06:00:11 +00:00
|
|
|
{
|
|
|
|
/* Stop if we're a zombie or need a soft hangup */
|
2006-04-06 15:40:12 +00:00
|
|
|
if (ast_test_flag(c, AST_FLAG_ZOMBIE) || ast_check_hangup(c))
|
2003-02-23 06:00:11 +00:00
|
|
|
return -1;
|
2007-09-04 13:47:33 +00:00
|
|
|
|
|
|
|
/* Only look for the end of DTMF, don't bother with the beginning and don't emulate things */
|
|
|
|
ast_set_flag(c, AST_FLAG_END_DTMF_ONLY);
|
|
|
|
|
2003-02-23 06:00:11 +00:00
|
|
|
/* Wait for a digit, no more than ms milliseconds total. */
|
2007-08-14 13:46:58 +00:00
|
|
|
|
2006-04-16 08:01:10 +00:00
|
|
|
while (ms) {
|
|
|
|
struct ast_channel *rchan;
|
2007-08-14 13:46:58 +00:00
|
|
|
int outfd=-1;
|
2006-04-16 08:01:10 +00:00
|
|
|
|
2005-04-22 01:40:23 +00:00
|
|
|
errno = 0;
|
2004-04-12 16:25:34 +00:00
|
|
|
rchan = ast_waitfor_nandfds(&c, 1, &cmdfd, (cmdfd > -1) ? 1 : 0, NULL, &outfd, &ms);
|
2007-08-14 13:46:58 +00:00
|
|
|
|
2006-04-16 08:01:10 +00:00
|
|
|
if (!rchan && outfd < 0 && ms) {
|
2005-04-22 01:40:23 +00:00
|
|
|
if (errno == 0 || errno == EINTR)
|
2005-03-22 22:44:55 +00:00
|
|
|
continue;
|
2004-04-12 16:25:34 +00:00
|
|
|
ast_log(LOG_WARNING, "Wait failed (%s)\n", strerror(errno));
|
2007-09-04 13:47:33 +00:00
|
|
|
ast_clear_flag(c, AST_FLAG_END_DTMF_ONLY);
|
2004-04-12 16:25:34 +00:00
|
|
|
return -1;
|
|
|
|
} else if (outfd > -1) {
|
|
|
|
/* The FD we were watching has something waiting */
|
2007-08-14 13:46:58 +00:00
|
|
|
ast_log(LOG_WARNING, "The FD we were waiting for has something waiting. Waitfordigit returning numeric 1\n");
|
2007-09-04 13:47:33 +00:00
|
|
|
ast_clear_flag(c, AST_FLAG_END_DTMF_ONLY);
|
2004-04-12 16:25:34 +00:00
|
|
|
return 1;
|
2003-02-23 06:00:11 +00:00
|
|
|
} else if (rchan) {
|
2006-04-16 08:01:10 +00:00
|
|
|
int res;
|
|
|
|
struct ast_frame *f = ast_read(c);
|
|
|
|
if (!f)
|
2004-04-12 16:25:34 +00:00
|
|
|
return -1;
|
|
|
|
|
2007-01-23 00:11:32 +00:00
|
|
|
switch (f->frametype) {
|
2007-07-12 17:17:56 +00:00
|
|
|
case AST_FRAME_DTMF_BEGIN:
|
|
|
|
break;
|
|
|
|
case AST_FRAME_DTMF_END:
|
2004-04-12 16:25:34 +00:00
|
|
|
res = f->subclass;
|
2003-02-23 06:00:11 +00:00
|
|
|
ast_frfree(f);
|
2007-09-04 13:47:33 +00:00
|
|
|
ast_clear_flag(c, AST_FLAG_END_DTMF_ONLY);
|
2004-04-12 16:25:34 +00:00
|
|
|
return res;
|
|
|
|
case AST_FRAME_CONTROL:
|
2007-01-23 00:11:32 +00:00
|
|
|
switch (f->subclass) {
|
2004-04-12 16:25:34 +00:00
|
|
|
case AST_CONTROL_HANGUP:
|
|
|
|
ast_frfree(f);
|
2007-09-04 13:47:33 +00:00
|
|
|
ast_clear_flag(c, AST_FLAG_END_DTMF_ONLY);
|
2004-04-12 16:25:34 +00:00
|
|
|
return -1;
|
|
|
|
case AST_CONTROL_RINGING:
|
|
|
|
case AST_CONTROL_ANSWER:
|
2008-03-07 19:33:09 +00:00
|
|
|
case AST_CONTROL_SRCUPDATE:
|
2004-04-12 16:25:34 +00:00
|
|
|
/* Unimportant */
|
|
|
|
break;
|
|
|
|
default:
|
|
|
|
ast_log(LOG_WARNING, "Unexpected control subclass '%d'\n", f->subclass);
|
2006-09-01 16:34:57 +00:00
|
|
|
break;
|
2004-04-12 16:25:34 +00:00
|
|
|
}
|
2006-09-01 16:34:57 +00:00
|
|
|
break;
|
2004-04-12 16:25:34 +00:00
|
|
|
case AST_FRAME_VOICE:
|
|
|
|
/* Write audio if appropriate */
|
2008-11-02 18:52:13 +00:00
|
|
|
if (audiofd > -1) {
|
|
|
|
if (write(audiofd, f->data.ptr, f->datalen) < 0) {
|
|
|
|
ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
|
|
|
|
}
|
|
|
|
}
|
2006-08-31 01:59:02 +00:00
|
|
|
default:
|
|
|
|
/* Ignore */
|
|
|
|
break;
|
2004-04-12 16:25:34 +00:00
|
|
|
}
|
|
|
|
ast_frfree(f);
|
2003-02-23 06:00:11 +00:00
|
|
|
}
|
|
|
|
}
|
2007-09-04 13:47:33 +00:00
|
|
|
|
|
|
|
ast_clear_flag(c, AST_FLAG_END_DTMF_ONLY);
|
|
|
|
|
2005-01-15 23:48:12 +00:00
|
|
|
return 0; /* Time is up */
|
2003-02-23 06:00:11 +00:00
|
|
|
}
|
|
|
|
|
2007-04-10 23:55:26 +00:00
|
|
|
static void send_dtmf_event(const struct ast_channel *chan, const char *direction, const char digit, const char *begin, const char *end)
|
|
|
|
{
|
|
|
|
manager_event(EVENT_FLAG_DTMF,
|
|
|
|
"DTMF",
|
|
|
|
"Channel: %s\r\n"
|
|
|
|
"Uniqueid: %s\r\n"
|
|
|
|
"Digit: %c\r\n"
|
|
|
|
"Direction: %s\r\n"
|
|
|
|
"Begin: %s\r\n"
|
|
|
|
"End: %s\r\n",
|
|
|
|
chan->name, chan->uniqueid, digit, direction, begin, end);
|
|
|
|
}
|
|
|
|
|
2007-06-27 17:34:26 +00:00
|
|
|
static void ast_read_generator_actions(struct ast_channel *chan, struct ast_frame *f)
|
|
|
|
{
|
|
|
|
if (chan->generatordata && !ast_internal_timing_enabled(chan)) {
|
|
|
|
void *tmp = chan->generatordata;
|
2008-04-14 15:01:36 +00:00
|
|
|
int (*generate)(struct ast_channel *chan, void *tmp, int datalen, int samples) = NULL;
|
2007-06-27 17:34:26 +00:00
|
|
|
int res;
|
Add some fixes that I made in regards to wideband codec handling to get
G.722 music on hold working for me.
(issue #12164, reported by milazzo and jsmith, patches by me)
res/res_musiconhold.c:
- I moved a single line so that the sample queue update happened before
ast_write(). The reason that this was a bug is that the G.722 frame
originally says it has 320 samples in it (which is correct). However,
when the frame is written to a channel that uses RTP, main/rtp.c modifies
the frame to cut the number of samples in half before it sends it on
the wire. This is to account for the stupid incorrect G.722 spec that
makes it so we have to lie about the number of samples with RTP. I should
probably go and re-work the RTP code so it doesn't modify the frame so
that a bug like this won't happen in the future. However, this change to
MOH is harmless.
main/channel.c:
- I made two fixes in regards to generator timing. Generators use samples
for timing. However, this code assumed 8 kHz samples. In one case, it was
a hard coded 160 samples, that is now written as the sample rate / 50. The
other place was dealing with timing a generator based on frames coming from
the other direction. However, that would have only worked if the sample
rates for the formats in both directions were the same. The code now takes
into account that the sample rates may differ, and scales the generator
samples accordingly.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110268 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-20 17:41:22 +00:00
|
|
|
int samples;
|
2007-06-27 17:34:26 +00:00
|
|
|
|
|
|
|
if (chan->timingfunc) {
|
2008-04-15 15:20:47 +00:00
|
|
|
ast_debug(1, "Generator got voice, switching to phase locked mode\n");
|
2007-06-27 17:34:26 +00:00
|
|
|
ast_settimeout(chan, 0, NULL, NULL);
|
|
|
|
}
|
|
|
|
|
|
|
|
chan->generatordata = NULL; /* reset, to let writes go through */
|
Add some fixes that I made in regards to wideband codec handling to get
G.722 music on hold working for me.
(issue #12164, reported by milazzo and jsmith, patches by me)
res/res_musiconhold.c:
- I moved a single line so that the sample queue update happened before
ast_write(). The reason that this was a bug is that the G.722 frame
originally says it has 320 samples in it (which is correct). However,
when the frame is written to a channel that uses RTP, main/rtp.c modifies
the frame to cut the number of samples in half before it sends it on
the wire. This is to account for the stupid incorrect G.722 spec that
makes it so we have to lie about the number of samples with RTP. I should
probably go and re-work the RTP code so it doesn't modify the frame so
that a bug like this won't happen in the future. However, this change to
MOH is harmless.
main/channel.c:
- I made two fixes in regards to generator timing. Generators use samples
for timing. However, this code assumed 8 kHz samples. In one case, it was
a hard coded 160 samples, that is now written as the sample rate / 50. The
other place was dealing with timing a generator based on frames coming from
the other direction. However, that would have only worked if the sample
rates for the formats in both directions were the same. The code now takes
into account that the sample rates may differ, and scales the generator
samples accordingly.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110268 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-20 17:41:22 +00:00
|
|
|
|
|
|
|
if (f->subclass != chan->writeformat) {
|
|
|
|
float factor;
|
|
|
|
factor = ((float) ast_format_rate(chan->writeformat)) / ((float) ast_format_rate(f->subclass));
|
|
|
|
samples = (int) ( ((float) f->samples) * factor );
|
|
|
|
} else {
|
|
|
|
samples = f->samples;
|
|
|
|
}
|
2008-04-14 15:01:36 +00:00
|
|
|
|
|
|
|
if (chan->generator->generate) {
|
|
|
|
generate = chan->generator->generate;
|
|
|
|
}
|
2008-04-07 16:12:30 +00:00
|
|
|
/* This unlock is here based on two assumptions that hold true at this point in the
|
|
|
|
* code. 1) this function is only called from within __ast_read() and 2) all generators
|
|
|
|
* call ast_write() in their generate callback.
|
|
|
|
*
|
|
|
|
* The reason this is added is so that when ast_write is called, the lock that occurs
|
|
|
|
* there will not recursively lock the channel. Doing this will cause intended deadlock
|
|
|
|
* avoidance not to work in deeper functions
|
|
|
|
*/
|
|
|
|
ast_channel_unlock(chan);
|
2008-04-22 20:25:56 +00:00
|
|
|
res = generate(chan, tmp, f->datalen, samples);
|
2008-04-07 16:12:30 +00:00
|
|
|
ast_channel_lock(chan);
|
2007-06-27 17:34:26 +00:00
|
|
|
chan->generatordata = tmp;
|
|
|
|
if (res) {
|
2008-04-15 15:20:47 +00:00
|
|
|
ast_debug(1, "Auto-deactivating generator\n");
|
2007-06-27 17:34:26 +00:00
|
|
|
ast_deactivate_generator(chan);
|
|
|
|
}
|
|
|
|
|
|
|
|
} else if (f->frametype == AST_FRAME_CNG) {
|
|
|
|
if (chan->generator && !chan->timingfunc && (chan->timingfd > -1)) {
|
2008-04-15 15:20:47 +00:00
|
|
|
ast_debug(1, "Generator got CNG, switching to timed mode\n");
|
2008-06-13 12:45:50 +00:00
|
|
|
ast_settimeout(chan, 50, generator_force, chan);
|
2007-06-27 17:34:26 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2005-12-20 13:07:02 +00:00
|
|
|
static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio)
|
1999-12-04 21:35:07 +00:00
|
|
|
{
|
2006-04-16 13:27:58 +00:00
|
|
|
struct ast_frame *f = NULL; /* the return value */
|
2002-06-21 01:40:13 +00:00
|
|
|
int blah;
|
2005-03-31 19:07:27 +00:00
|
|
|
int prestate;
|
2008-04-24 22:16:48 +00:00
|
|
|
int count = 0, cause = 0;
|
2006-04-16 13:27:58 +00:00
|
|
|
|
2006-04-16 17:41:05 +00:00
|
|
|
/* this function is very long so make sure there is only one return
|
2007-11-05 23:31:00 +00:00
|
|
|
* point at the end (there are only two exceptions to this).
|
2006-04-16 17:41:05 +00:00
|
|
|
*/
|
2007-11-05 23:31:00 +00:00
|
|
|
while(ast_channel_trylock(chan)) {
|
|
|
|
if(count++ > 10)
|
|
|
|
/*cannot goto done since the channel is not locked*/
|
|
|
|
return &ast_null_frame;
|
|
|
|
usleep(1);
|
|
|
|
}
|
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
|
if (chan->masq) {
|
2006-08-31 01:59:02 +00:00
|
|
|
if (ast_do_masquerade(chan))
|
2001-05-07 00:43:32 +00:00
|
|
|
ast_log(LOG_WARNING, "Failed to perform masquerade\n");
|
2006-08-31 01:59:02 +00:00
|
|
|
else
|
2006-01-31 17:18:58 +00:00
|
|
|
f = &ast_null_frame;
|
2006-04-16 17:41:05 +00:00
|
|
|
goto done;
|
2001-05-07 00:43:32 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
/* Stop if we're a zombie or need a soft hangup */
|
2004-12-07 20:38:43 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_ZOMBIE) || ast_check_hangup(chan)) {
|
2002-09-10 20:48:20 +00:00
|
|
|
if (chan->generator)
|
2004-04-07 14:43:18 +00:00
|
|
|
ast_deactivate_generator(chan);
|
2006-04-16 17:41:05 +00:00
|
|
|
goto done;
|
2001-05-07 00:43:32 +00:00
|
|
|
}
|
2005-03-31 03:00:37 +00:00
|
|
|
prestate = chan->_state;
|
2002-06-21 01:40:13 +00:00
|
|
|
|
2006-08-31 01:59:02 +00:00
|
|
|
if (!ast_test_flag(chan, AST_FLAG_DEFER_DTMF | AST_FLAG_EMULATE_DTMF | AST_FLAG_IN_DTMF) &&
|
2007-04-24 19:03:16 +00:00
|
|
|
!ast_strlen_zero(chan->dtmfq) &&
|
|
|
|
(ast_tvzero(chan->dtmf_tv) || ast_tvdiff_ms(ast_tvnow(), chan->dtmf_tv) > AST_MIN_DTMF_GAP) ) {
|
2004-04-27 21:21:57 +00:00
|
|
|
/* We have DTMF that has been deferred. Return it now */
|
2001-09-19 14:40:16 +00:00
|
|
|
chan->dtmff.subclass = chan->dtmfq[0];
|
2006-04-16 17:41:05 +00:00
|
|
|
/* Drop first digit from the buffer */
|
2001-09-19 14:40:16 +00:00
|
|
|
memmove(chan->dtmfq, chan->dtmfq + 1, sizeof(chan->dtmfq) - 1);
|
2006-04-16 17:41:05 +00:00
|
|
|
f = &chan->dtmff;
|
2007-07-02 14:39:49 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_END_DTMF_ONLY)) {
|
|
|
|
ast_log(LOG_DTMF, "DTMF end emulation of '%c' queued on %s\n", f->subclass, chan->name);
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
chan->dtmff.frametype = AST_FRAME_DTMF_END;
|
2007-07-02 14:39:49 +00:00
|
|
|
} else {
|
|
|
|
ast_log(LOG_DTMF, "DTMF begin emulation of '%c' with duration %d queued on %s\n", f->subclass, AST_DEFAULT_EMULATE_DTMF_DURATION, chan->name);
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
chan->dtmff.frametype = AST_FRAME_DTMF_BEGIN;
|
|
|
|
ast_set_flag(chan, AST_FLAG_EMULATE_DTMF);
|
|
|
|
chan->emulate_dtmf_digit = f->subclass;
|
|
|
|
chan->emulate_dtmf_duration = AST_DEFAULT_EMULATE_DTMF_DURATION;
|
|
|
|
}
|
2007-04-24 19:03:16 +00:00
|
|
|
chan->dtmf_tv = ast_tvnow();
|
2006-04-16 17:41:05 +00:00
|
|
|
goto done;
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
|
|
|
|
2002-06-21 01:40:13 +00:00
|
|
|
/* Read and ignore anything on the alertpipe, but read only
|
|
|
|
one sizeof(blah) per frame that we send from it */
|
2008-06-12 15:14:37 +00:00
|
|
|
if (chan->alertpipe[0] > -1) {
|
|
|
|
int flags = fcntl(chan->alertpipe[0], F_GETFL);
|
|
|
|
/* For some odd reason, the alertpipe occasionally loses nonblocking status,
|
|
|
|
* which immediately causes a deadlock scenario. Detect and prevent this. */
|
|
|
|
if ((flags & O_NONBLOCK) == 0) {
|
|
|
|
ast_log(LOG_ERROR, "Alertpipe on channel %s lost O_NONBLOCK?!!\n", chan->name);
|
|
|
|
if (fcntl(chan->alertpipe[0], F_SETFL, flags | O_NONBLOCK) < 0) {
|
|
|
|
ast_log(LOG_WARNING, "Unable to set alertpipe nonblocking! (%d: %s)\n", errno, strerror(errno));
|
|
|
|
f = &ast_null_frame;
|
|
|
|
goto done;
|
|
|
|
}
|
|
|
|
}
|
2008-11-02 18:52:13 +00:00
|
|
|
if (read(chan->alertpipe[0], &blah, sizeof(blah)) < 0) {
|
2008-11-05 16:11:11 +00:00
|
|
|
if (errno != EINTR && errno != EAGAIN)
|
|
|
|
ast_log(LOG_WARNING, "read() failed: %s\n", strerror(errno));
|
2008-11-02 18:52:13 +00:00
|
|
|
}
|
2008-06-12 15:14:37 +00:00
|
|
|
}
|
2006-04-16 17:41:05 +00:00
|
|
|
|
2008-06-16 12:48:11 +00:00
|
|
|
if (chan->timingfd > -1 && chan->fdno == AST_TIMING_FD) {
|
|
|
|
enum ast_timing_event res;
|
2006-04-16 13:27:58 +00:00
|
|
|
|
2004-12-07 20:38:43 +00:00
|
|
|
ast_clear_flag(chan, AST_FLAG_EXCEPTION);
|
2008-06-16 12:48:11 +00:00
|
|
|
|
|
|
|
res = ast_timer_get_event(chan->timingfd);
|
|
|
|
|
|
|
|
switch (res) {
|
|
|
|
case AST_TIMING_EVENT_EXPIRED:
|
|
|
|
ast_timer_ack(chan->timingfd, 1);
|
|
|
|
|
2006-04-16 13:27:58 +00:00
|
|
|
if (chan->timingfunc) {
|
2008-01-28 17:21:24 +00:00
|
|
|
/* save a copy of func/data before unlocking the channel */
|
|
|
|
int (*func)(const void *) = chan->timingfunc;
|
|
|
|
void *data = chan->timingdata;
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
func(data);
|
2004-03-19 04:23:20 +00:00
|
|
|
} else {
|
2008-06-16 12:48:11 +00:00
|
|
|
ast_timer_set_rate(chan->timingfd, 0);
|
2008-01-28 17:21:24 +00:00
|
|
|
ast_channel_unlock(chan);
|
2004-03-19 04:23:20 +00:00
|
|
|
}
|
2008-06-16 12:48:11 +00:00
|
|
|
|
2006-04-16 17:41:05 +00:00
|
|
|
/* cannot 'goto done' because the channel is already unlocked */
|
2006-01-31 17:18:58 +00:00
|
|
|
return &ast_null_frame;
|
2008-06-16 12:48:11 +00:00
|
|
|
|
|
|
|
case AST_TIMING_EVENT_CONTINUOUS:
|
|
|
|
if (AST_LIST_EMPTY(&chan->readq) ||
|
|
|
|
!AST_LIST_NEXT(AST_LIST_FIRST(&chan->readq), frame_list)) {
|
|
|
|
ast_timer_disable_continuous(chan->timingfd);
|
|
|
|
}
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
|
|
|
|
} else if (chan->fds[AST_GENERATOR_FD] > -1 && chan->fdno == AST_GENERATOR_FD) {
|
2006-04-16 17:41:05 +00:00
|
|
|
/* if the AST_GENERATOR_FD is set, call the generator with args
|
|
|
|
* set to -1 so it can do whatever it needs to.
|
|
|
|
*/
|
2006-01-30 03:13:33 +00:00
|
|
|
void *tmp = chan->generatordata;
|
|
|
|
chan->generatordata = NULL; /* reset to let ast_write get through */
|
|
|
|
chan->generator->generate(chan, tmp, -1, -1);
|
|
|
|
chan->generatordata = tmp;
|
2006-04-16 17:41:05 +00:00
|
|
|
f = &ast_null_frame;
|
|
|
|
goto done;
|
2006-01-30 03:13:33 +00:00
|
|
|
}
|
|
|
|
|
2002-06-21 01:40:13 +00:00
|
|
|
/* Check for pending read queue */
|
2006-08-29 20:50:36 +00:00
|
|
|
if (!AST_LIST_EMPTY(&chan->readq)) {
|
|
|
|
f = AST_LIST_REMOVE_HEAD(&chan->readq, frame_list);
|
2002-06-21 01:40:13 +00:00
|
|
|
/* Interpret hangup and return NULL */
|
2006-05-09 07:51:55 +00:00
|
|
|
/* XXX why not the same for frames from the channel ? */
|
2006-04-14 20:47:06 +00:00
|
|
|
if (f->frametype == AST_FRAME_CONTROL && f->subclass == AST_CONTROL_HANGUP) {
|
2008-05-22 16:29:54 +00:00
|
|
|
cause = f->data.uint32;
|
2003-11-07 03:48:03 +00:00
|
|
|
ast_frfree(f);
|
2002-06-21 01:40:13 +00:00
|
|
|
f = NULL;
|
2003-11-07 03:48:03 +00:00
|
|
|
}
|
2002-06-21 01:40:13 +00:00
|
|
|
} else {
|
|
|
|
chan->blocker = pthread_self();
|
2004-12-07 20:38:43 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_EXCEPTION)) {
|
2006-04-06 15:40:12 +00:00
|
|
|
if (chan->tech->exception)
|
2005-03-04 06:47:24 +00:00
|
|
|
f = chan->tech->exception(chan);
|
2003-02-05 21:18:09 +00:00
|
|
|
else {
|
|
|
|
ast_log(LOG_WARNING, "Exception flag set on '%s', but no exception handler\n", chan->name);
|
2006-01-31 17:18:58 +00:00
|
|
|
f = &ast_null_frame;
|
2003-02-05 21:18:09 +00:00
|
|
|
}
|
2002-06-21 01:40:13 +00:00
|
|
|
/* Clear the exception flag */
|
2004-12-07 20:38:43 +00:00
|
|
|
ast_clear_flag(chan, AST_FLAG_EXCEPTION);
|
2006-04-16 13:58:43 +00:00
|
|
|
} else if (chan->tech->read)
|
|
|
|
f = chan->tech->read(chan);
|
|
|
|
else
|
|
|
|
ast_log(LOG_WARNING, "No read routine on channel %s\n", chan->name);
|
2002-06-21 01:40:13 +00:00
|
|
|
}
|
|
|
|
|
2006-01-29 05:07:04 +00:00
|
|
|
if (f) {
|
|
|
|
/* if the channel driver returned more than one frame, stuff the excess
|
2006-08-29 20:50:36 +00:00
|
|
|
into the readq for the next ast_read call (note that we can safely assume
|
|
|
|
that the readq is empty, because otherwise we would not have called into
|
|
|
|
the channel driver and f would be only a single frame)
|
2006-01-29 05:07:04 +00:00
|
|
|
*/
|
2006-08-29 20:50:36 +00:00
|
|
|
if (AST_LIST_NEXT(f, frame_list)) {
|
|
|
|
AST_LIST_HEAD_SET_NOLOCK(&chan->readq, AST_LIST_NEXT(f, frame_list));
|
|
|
|
AST_LIST_NEXT(f, frame_list) = NULL;
|
2006-01-29 05:07:04 +00:00
|
|
|
}
|
2003-02-05 21:18:09 +00:00
|
|
|
|
2006-01-29 05:07:04 +00:00
|
|
|
switch (f->frametype) {
|
|
|
|
case AST_FRAME_CONTROL:
|
|
|
|
if (f->subclass == AST_CONTROL_ANSWER) {
|
2006-05-09 15:01:10 +00:00
|
|
|
if (!ast_test_flag(chan, AST_FLAG_OUTGOING)) {
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Ignoring answer on an inbound call!\n");
|
2006-05-16 20:08:29 +00:00
|
|
|
ast_frfree(f);
|
2006-05-09 15:01:10 +00:00
|
|
|
f = &ast_null_frame;
|
|
|
|
} else if (prestate == AST_STATE_UP) {
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Dropping duplicate answer!\n");
|
2006-05-16 20:08:29 +00:00
|
|
|
ast_frfree(f);
|
2006-01-31 17:18:58 +00:00
|
|
|
f = &ast_null_frame;
|
2006-05-09 15:01:10 +00:00
|
|
|
} else {
|
|
|
|
/* Answer the CDR */
|
|
|
|
ast_setstate(chan, AST_STATE_UP);
|
2008-07-03 17:16:44 +00:00
|
|
|
/* removed a call to ast_cdr_answer(chan->cdr) from here. */
|
2006-01-29 05:07:04 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
break;
|
2006-08-31 01:59:02 +00:00
|
|
|
case AST_FRAME_DTMF_END:
|
2007-04-10 23:55:26 +00:00
|
|
|
send_dtmf_event(chan, "Received", f->subclass, "No", "Yes");
|
2007-02-10 09:21:22 +00:00
|
|
|
ast_log(LOG_DTMF, "DTMF end '%c' received on %s, duration %ld ms\n", f->subclass, chan->name, f->len);
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
/* Queue it up if DTMF is deffered, or if DTMF emulation is forced.
|
|
|
|
* However, only let emulation be forced if the other end cares about BEGIN frames */
|
|
|
|
if ( ast_test_flag(chan, AST_FLAG_DEFER_DTMF) ||
|
|
|
|
(ast_test_flag(chan, AST_FLAG_EMULATE_DTMF) && !ast_test_flag(chan, AST_FLAG_END_DTMF_ONLY)) ) {
|
2007-08-06 21:43:09 +00:00
|
|
|
if (strlen(chan->dtmfq) < sizeof(chan->dtmfq) - 2) {
|
|
|
|
ast_log(LOG_DTMF, "DTMF end '%c' put into dtmf queue on %s\n", f->subclass, chan->name);
|
2006-01-29 05:07:04 +00:00
|
|
|
chan->dtmfq[strlen(chan->dtmfq)] = f->subclass;
|
2007-08-06 21:43:09 +00:00
|
|
|
} else
|
2006-01-29 05:07:04 +00:00
|
|
|
ast_log(LOG_WARNING, "Dropping deferred DTMF digits on %s\n", chan->name);
|
2006-05-16 20:08:29 +00:00
|
|
|
ast_frfree(f);
|
2006-01-31 17:18:58 +00:00
|
|
|
f = &ast_null_frame;
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
} else if (!ast_test_flag(chan, AST_FLAG_IN_DTMF | AST_FLAG_END_DTMF_ONLY)) {
|
2007-04-24 19:03:16 +00:00
|
|
|
if (!ast_tvzero(chan->dtmf_tv) &&
|
|
|
|
ast_tvdiff_ms(ast_tvnow(), chan->dtmf_tv) < AST_MIN_DTMF_GAP) {
|
|
|
|
/* If it hasn't been long enough, defer this digit */
|
2007-08-06 21:43:09 +00:00
|
|
|
if (strlen(chan->dtmfq) < sizeof(chan->dtmfq) - 2) {
|
|
|
|
ast_log(LOG_DTMF, "DTMF end '%c' put into dtmf queue on %s\n", f->subclass, chan->name);
|
2007-04-24 19:03:16 +00:00
|
|
|
chan->dtmfq[strlen(chan->dtmfq)] = f->subclass;
|
2007-08-06 21:43:09 +00:00
|
|
|
} else
|
2007-04-24 19:03:16 +00:00
|
|
|
ast_log(LOG_WARNING, "Dropping deferred DTMF digits on %s\n", chan->name);
|
|
|
|
ast_frfree(f);
|
|
|
|
f = &ast_null_frame;
|
|
|
|
} else {
|
|
|
|
/* There was no begin, turn this into a begin and send the end later */
|
|
|
|
f->frametype = AST_FRAME_DTMF_BEGIN;
|
|
|
|
ast_set_flag(chan, AST_FLAG_EMULATE_DTMF);
|
|
|
|
chan->emulate_dtmf_digit = f->subclass;
|
|
|
|
chan->dtmf_tv = ast_tvnow();
|
Merged revisions 62942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines
Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).
This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end. This is fixed,
along with a couple other little improvements.
* When chan_zap is in the middle of playing a digit to a channel, it feeds
back null frames, not voice frames. So, I have modified ast_read to check
the timing on emulated DTMF when it receives null frames, in addition to
where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits. If there was
no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
frames that pass through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62943 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03 15:23:44 +00:00
|
|
|
if (f->len) {
|
|
|
|
if (f->len > AST_MIN_DTMF_DURATION)
|
|
|
|
chan->emulate_dtmf_duration = f->len;
|
|
|
|
else
|
|
|
|
chan->emulate_dtmf_duration = AST_MIN_DTMF_DURATION;
|
|
|
|
} else
|
|
|
|
chan->emulate_dtmf_duration = AST_DEFAULT_EMULATE_DTMF_DURATION;
|
2007-07-25 21:53:35 +00:00
|
|
|
ast_log(LOG_DTMF, "DTMF begin emulation of '%c' with duration %u queued on %s\n", f->subclass, chan->emulate_dtmf_duration, chan->name);
|
2007-04-24 19:03:16 +00:00
|
|
|
}
|
2007-09-06 22:34:44 +00:00
|
|
|
if (chan->audiohooks) {
|
|
|
|
struct ast_frame *old_frame = f;
|
2007-12-26 18:19:46 +00:00
|
|
|
/*!
|
|
|
|
* \todo XXX It is possible to write a digit to the audiohook twice
|
|
|
|
* if the digit was originally read while the channel was in autoservice. */
|
2007-08-08 19:30:52 +00:00
|
|
|
f = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_READ, f);
|
2007-09-06 22:34:44 +00:00
|
|
|
if (old_frame != f)
|
|
|
|
ast_frfree(old_frame);
|
|
|
|
}
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
} else {
|
2007-04-24 19:03:16 +00:00
|
|
|
struct timeval now = ast_tvnow();
|
2007-07-26 23:20:25 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_IN_DTMF)) {
|
2007-08-06 21:43:09 +00:00
|
|
|
ast_log(LOG_DTMF, "DTMF end accepted with begin '%c' on %s\n", f->subclass, chan->name);
|
2007-07-26 23:20:25 +00:00
|
|
|
ast_clear_flag(chan, AST_FLAG_IN_DTMF);
|
|
|
|
if (!f->len)
|
|
|
|
f->len = ast_tvdiff_ms(now, chan->dtmf_tv);
|
2007-08-06 21:43:09 +00:00
|
|
|
} else if (!f->len) {
|
|
|
|
ast_log(LOG_DTMF, "DTMF end accepted without begin '%c' on %s\n", f->subclass, chan->name);
|
2007-07-26 23:20:25 +00:00
|
|
|
f->len = AST_MIN_DTMF_DURATION;
|
2007-08-06 21:43:09 +00:00
|
|
|
}
|
2008-06-09 16:37:08 +00:00
|
|
|
if (f->len < AST_MIN_DTMF_DURATION && !ast_test_flag(chan, AST_FLAG_END_DTMF_ONLY)) {
|
2007-08-06 21:43:09 +00:00
|
|
|
ast_log(LOG_DTMF, "DTMF end '%c' has duration %ld but want minimum %d, emulating on %s\n", f->subclass, f->len, AST_MIN_DTMF_DURATION, chan->name);
|
2007-04-23 17:58:15 +00:00
|
|
|
ast_set_flag(chan, AST_FLAG_EMULATE_DTMF);
|
|
|
|
chan->emulate_dtmf_digit = f->subclass;
|
2007-04-24 19:03:16 +00:00
|
|
|
chan->emulate_dtmf_duration = AST_MIN_DTMF_DURATION - f->len;
|
2007-10-22 15:57:02 +00:00
|
|
|
ast_frfree(f);
|
2007-04-23 17:58:15 +00:00
|
|
|
f = &ast_null_frame;
|
2007-08-06 21:43:09 +00:00
|
|
|
} else {
|
|
|
|
ast_log(LOG_DTMF, "DTMF end passthrough '%c' on %s\n", f->subclass, chan->name);
|
2008-06-09 16:37:08 +00:00
|
|
|
if (f->len < AST_MIN_DTMF_DURATION) {
|
|
|
|
f->len = AST_MIN_DTMF_DURATION;
|
|
|
|
}
|
2007-04-24 19:03:16 +00:00
|
|
|
chan->dtmf_tv = now;
|
2007-08-06 21:43:09 +00:00
|
|
|
}
|
2007-09-06 22:34:44 +00:00
|
|
|
if (chan->audiohooks) {
|
|
|
|
struct ast_frame *old_frame = f;
|
2007-08-08 19:30:52 +00:00
|
|
|
f = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_READ, f);
|
2007-09-06 22:34:44 +00:00
|
|
|
if (old_frame != f)
|
|
|
|
ast_frfree(old_frame);
|
|
|
|
}
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
}
|
2006-01-29 05:07:04 +00:00
|
|
|
break;
|
|
|
|
case AST_FRAME_DTMF_BEGIN:
|
2007-04-10 23:55:26 +00:00
|
|
|
send_dtmf_event(chan, "Received", f->subclass, "Yes", "No");
|
2006-01-29 05:07:04 +00:00
|
|
|
ast_log(LOG_DTMF, "DTMF begin '%c' received on %s\n", f->subclass, chan->name);
|
2008-02-18 23:58:38 +00:00
|
|
|
if ( ast_test_flag(chan, AST_FLAG_DEFER_DTMF | AST_FLAG_END_DTMF_ONLY | AST_FLAG_EMULATE_DTMF) ||
|
2007-04-24 19:03:16 +00:00
|
|
|
(!ast_tvzero(chan->dtmf_tv) &&
|
|
|
|
ast_tvdiff_ms(ast_tvnow(), chan->dtmf_tv) < AST_MIN_DTMF_GAP) ) {
|
2007-10-22 14:28:07 +00:00
|
|
|
ast_log(LOG_DTMF, "DTMF begin ignored '%c' on %s\n", f->subclass, chan->name);
|
2006-08-31 01:59:02 +00:00
|
|
|
ast_frfree(f);
|
|
|
|
f = &ast_null_frame;
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
} else {
|
2006-08-31 01:59:02 +00:00
|
|
|
ast_set_flag(chan, AST_FLAG_IN_DTMF);
|
2007-04-24 19:03:16 +00:00
|
|
|
chan->dtmf_tv = ast_tvnow();
|
2007-08-06 21:43:09 +00:00
|
|
|
ast_log(LOG_DTMF, "DTMF begin passthrough '%c' on %s\n", f->subclass, chan->name);
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
}
|
2006-01-29 05:07:04 +00:00
|
|
|
break;
|
Merged revisions 62942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines
Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).
This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end. This is fixed,
along with a couple other little improvements.
* When chan_zap is in the middle of playing a digit to a channel, it feeds
back null frames, not voice frames. So, I have modified ast_read to check
the timing on emulated DTMF when it receives null frames, in addition to
where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits. If there was
no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
frames that pass through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62943 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03 15:23:44 +00:00
|
|
|
case AST_FRAME_NULL:
|
2008-03-03 15:30:29 +00:00
|
|
|
/* The EMULATE_DTMF flag must be cleared here as opposed to when the duration
|
|
|
|
* is reached , because we want to make sure we pass at least one
|
|
|
|
* voice frame through before starting the next digit, to ensure a gap
|
|
|
|
* between DTMF digits. */
|
Merged revisions 62942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines
Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).
This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end. This is fixed,
along with a couple other little improvements.
* When chan_zap is in the middle of playing a digit to a channel, it feeds
back null frames, not voice frames. So, I have modified ast_read to check
the timing on emulated DTMF when it receives null frames, in addition to
where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits. If there was
no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
frames that pass through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62943 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03 15:23:44 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_EMULATE_DTMF)) {
|
|
|
|
struct timeval now = ast_tvnow();
|
2008-03-03 15:30:29 +00:00
|
|
|
if (!chan->emulate_dtmf_duration) {
|
|
|
|
ast_clear_flag(chan, AST_FLAG_EMULATE_DTMF);
|
|
|
|
chan->emulate_dtmf_digit = 0;
|
|
|
|
} else if (ast_tvdiff_ms(now, chan->dtmf_tv) >= chan->emulate_dtmf_duration) {
|
Merged revisions 62942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines
Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).
This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end. This is fixed,
along with a couple other little improvements.
* When chan_zap is in the middle of playing a digit to a channel, it feeds
back null frames, not voice frames. So, I have modified ast_read to check
the timing on emulated DTMF when it receives null frames, in addition to
where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits. If there was
no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
frames that pass through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62943 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03 15:23:44 +00:00
|
|
|
chan->emulate_dtmf_duration = 0;
|
2007-05-09 19:24:27 +00:00
|
|
|
ast_frfree(f);
|
|
|
|
f = &chan->dtmff;
|
Merged revisions 62942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines
Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).
This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end. This is fixed,
along with a couple other little improvements.
* When chan_zap is in the middle of playing a digit to a channel, it feeds
back null frames, not voice frames. So, I have modified ast_read to check
the timing on emulated DTMF when it receives null frames, in addition to
where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits. If there was
no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
frames that pass through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62943 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03 15:23:44 +00:00
|
|
|
f->frametype = AST_FRAME_DTMF_END;
|
|
|
|
f->subclass = chan->emulate_dtmf_digit;
|
|
|
|
f->len = ast_tvdiff_ms(now, chan->dtmf_tv);
|
|
|
|
chan->dtmf_tv = now;
|
|
|
|
ast_clear_flag(chan, AST_FLAG_EMULATE_DTMF);
|
|
|
|
chan->emulate_dtmf_digit = 0;
|
2007-07-02 14:39:49 +00:00
|
|
|
ast_log(LOG_DTMF, "DTMF end emulation of '%c' queued on %s\n", f->subclass, chan->name);
|
Merged revisions 62942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines
Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).
This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end. This is fixed,
along with a couple other little improvements.
* When chan_zap is in the middle of playing a digit to a channel, it feeds
back null frames, not voice frames. So, I have modified ast_read to check
the timing on emulated DTMF when it receives null frames, in addition to
where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits. If there was
no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
frames that pass through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62943 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03 15:23:44 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
break;
|
2006-01-29 05:07:04 +00:00
|
|
|
case AST_FRAME_VOICE:
|
Merged revisions 62942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines
Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).
This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end. This is fixed,
along with a couple other little improvements.
* When chan_zap is in the middle of playing a digit to a channel, it feeds
back null frames, not voice frames. So, I have modified ast_read to check
the timing on emulated DTMF when it receives null frames, in addition to
where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits. If there was
no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
frames that pass through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62943 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03 15:23:44 +00:00
|
|
|
/* The EMULATE_DTMF flag must be cleared here as opposed to when the duration
|
|
|
|
* is reached , because we want to make sure we pass at least one
|
2006-08-31 01:59:02 +00:00
|
|
|
* voice frame through before starting the next digit, to ensure a gap
|
|
|
|
* between DTMF digits. */
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_EMULATE_DTMF) && !chan->emulate_dtmf_duration) {
|
2006-08-31 01:59:02 +00:00
|
|
|
ast_clear_flag(chan, AST_FLAG_EMULATE_DTMF);
|
|
|
|
chan->emulate_dtmf_digit = 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (dropaudio || ast_test_flag(chan, AST_FLAG_IN_DTMF)) {
|
2007-06-27 17:34:26 +00:00
|
|
|
if (dropaudio)
|
|
|
|
ast_read_generator_actions(chan, f);
|
2006-01-29 05:07:04 +00:00
|
|
|
ast_frfree(f);
|
2006-01-31 17:18:58 +00:00
|
|
|
f = &ast_null_frame;
|
2007-06-27 17:34:26 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
if (ast_test_flag(chan, AST_FLAG_EMULATE_DTMF) && !ast_test_flag(chan, AST_FLAG_IN_DTMF)) {
|
Merged revisions 62942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines
Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).
This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end. This is fixed,
along with a couple other little improvements.
* When chan_zap is in the middle of playing a digit to a channel, it feeds
back null frames, not voice frames. So, I have modified ast_read to check
the timing on emulated DTMF when it receives null frames, in addition to
where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits. If there was
no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
frames that pass through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62943 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03 15:23:44 +00:00
|
|
|
struct timeval now = ast_tvnow();
|
|
|
|
if (ast_tvdiff_ms(now, chan->dtmf_tv) >= chan->emulate_dtmf_duration) {
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
chan->emulate_dtmf_duration = 0;
|
2007-05-09 19:24:27 +00:00
|
|
|
ast_frfree(f);
|
|
|
|
f = &chan->dtmff;
|
2006-08-31 01:59:02 +00:00
|
|
|
f->frametype = AST_FRAME_DTMF_END;
|
|
|
|
f->subclass = chan->emulate_dtmf_digit;
|
2007-04-24 19:03:16 +00:00
|
|
|
f->len = ast_tvdiff_ms(now, chan->dtmf_tv);
|
|
|
|
chan->dtmf_tv = now;
|
2007-09-06 22:34:44 +00:00
|
|
|
if (chan->audiohooks) {
|
|
|
|
struct ast_frame *old_frame = f;
|
2007-08-08 19:30:52 +00:00
|
|
|
f = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_READ, f);
|
2007-09-06 22:34:44 +00:00
|
|
|
if (old_frame != f)
|
|
|
|
ast_frfree(old_frame);
|
|
|
|
}
|
2007-07-02 14:39:49 +00:00
|
|
|
ast_log(LOG_DTMF, "DTMF end emulation of '%c' queued on %s\n", f->subclass, chan->name);
|
2006-08-31 01:59:02 +00:00
|
|
|
} else {
|
Merged revisions 62942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines
Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).
This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end. This is fixed,
along with a couple other little improvements.
* When chan_zap is in the middle of playing a digit to a channel, it feeds
back null frames, not voice frames. So, I have modified ast_read to check
the timing on emulated DTMF when it receives null frames, in addition to
where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits. If there was
no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
frames that pass through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62943 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03 15:23:44 +00:00
|
|
|
/* Drop voice frames while we're still in the middle of the digit */
|
2006-08-31 01:59:02 +00:00
|
|
|
ast_frfree(f);
|
|
|
|
f = &ast_null_frame;
|
|
|
|
}
|
2007-06-27 17:34:26 +00:00
|
|
|
} else if ((f->frametype == AST_FRAME_VOICE) && !(f->subclass & chan->nativeformats)) {
|
2008-06-19 17:02:54 +00:00
|
|
|
/* This frame is not one of the current native formats -- drop it on the floor */
|
|
|
|
char to[200];
|
2006-01-29 05:07:04 +00:00
|
|
|
ast_log(LOG_NOTICE, "Dropping incompatible voice frame on %s of format %s since our native format has changed to %s\n",
|
2008-06-19 17:02:54 +00:00
|
|
|
chan->name, ast_getformatname(f->subclass), ast_getformatname_multiple(to, sizeof(to), chan->nativeformats));
|
2006-01-29 05:07:04 +00:00
|
|
|
ast_frfree(f);
|
2006-01-31 17:18:58 +00:00
|
|
|
f = &ast_null_frame;
|
2007-06-27 17:34:26 +00:00
|
|
|
} else if ((f->frametype == AST_FRAME_VOICE)) {
|
2007-08-08 19:30:52 +00:00
|
|
|
/* Send frame to audiohooks if present */
|
2007-09-06 22:34:44 +00:00
|
|
|
if (chan->audiohooks) {
|
|
|
|
struct ast_frame *old_frame = f;
|
2007-08-08 19:30:52 +00:00
|
|
|
f = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_READ, f);
|
2007-09-06 22:34:44 +00:00
|
|
|
if (old_frame != f)
|
|
|
|
ast_frfree(old_frame);
|
|
|
|
}
|
2006-01-29 05:07:04 +00:00
|
|
|
if (chan->monitor && chan->monitor->read_stream ) {
|
2006-05-09 07:51:55 +00:00
|
|
|
/* XXX what does this do ? */
|
2003-08-28 20:02:10 +00:00
|
|
|
#ifndef MONITOR_CONSTANT_DELAY
|
2006-01-29 05:07:04 +00:00
|
|
|
int jump = chan->outsmpl - chan->insmpl - 4 * f->samples;
|
|
|
|
if (jump >= 0) {
|
2007-01-24 01:00:57 +00:00
|
|
|
jump = chan->outsmpl - chan->insmpl;
|
|
|
|
if (ast_seekstream(chan->monitor->read_stream, jump, SEEK_FORCECUR) == -1)
|
2006-01-29 05:07:04 +00:00
|
|
|
ast_log(LOG_WARNING, "Failed to perform seek in monitoring read stream, synchronization between the files may be broken\n");
|
2007-01-24 01:00:57 +00:00
|
|
|
chan->insmpl += jump + f->samples;
|
2006-01-29 05:07:04 +00:00
|
|
|
} else
|
|
|
|
chan->insmpl+= f->samples;
|
2003-08-28 20:02:10 +00:00
|
|
|
#else
|
2006-01-29 05:07:04 +00:00
|
|
|
int jump = chan->outsmpl - chan->insmpl;
|
|
|
|
if (jump - MONITOR_DELAY >= 0) {
|
|
|
|
if (ast_seekstream(chan->monitor->read_stream, jump - f->samples, SEEK_FORCECUR) == -1)
|
|
|
|
ast_log(LOG_WARNING, "Failed to perform seek in monitoring read stream, synchronization between the files may be broken\n");
|
|
|
|
chan->insmpl += jump;
|
|
|
|
} else
|
|
|
|
chan->insmpl += f->samples;
|
2003-08-28 20:02:10 +00:00
|
|
|
#endif
|
2006-01-29 05:07:04 +00:00
|
|
|
if (chan->monitor->state == AST_MONITOR_RUNNING) {
|
|
|
|
if (ast_writestream(chan->monitor->read_stream, f) < 0)
|
|
|
|
ast_log(LOG_WARNING, "Failed to write data to channel monitor read stream\n");
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2006-04-16 13:58:43 +00:00
|
|
|
if (chan->readtrans && (f = ast_translate(chan->readtrans, f, 1)) == NULL)
|
|
|
|
f = &ast_null_frame;
|
2007-06-27 17:34:26 +00:00
|
|
|
else
|
|
|
|
/* Run generator sitting on the line if timing device not available
|
|
|
|
* and synchronous generation of outgoing frames is necessary */
|
|
|
|
ast_read_generator_actions(chan, f);
|
2003-05-28 19:38:02 +00:00
|
|
|
}
|
2006-08-31 01:59:02 +00:00
|
|
|
default:
|
|
|
|
/* Just pass it on! */
|
|
|
|
break;
|
2001-03-12 03:04:51 +00:00
|
|
|
}
|
2006-01-29 05:07:04 +00:00
|
|
|
} else {
|
|
|
|
/* Make sure we always return NULL in the future */
|
2002-09-10 20:48:20 +00:00
|
|
|
chan->_softhangup |= AST_SOFTHANGUP_DEV;
|
2008-04-24 22:16:48 +00:00
|
|
|
if (cause)
|
|
|
|
chan->hangupcause = cause;
|
2002-09-10 20:48:20 +00:00
|
|
|
if (chan->generator)
|
2004-04-07 14:43:18 +00:00
|
|
|
ast_deactivate_generator(chan);
|
2008-07-03 17:16:44 +00:00
|
|
|
/* We no longer End the CDR here */
|
2002-06-21 01:40:13 +00:00
|
|
|
}
|
2006-01-29 05:07:04 +00:00
|
|
|
|
2005-05-14 23:26:37 +00:00
|
|
|
/* High bit prints debugging */
|
2006-04-21 14:49:21 +00:00
|
|
|
if (chan->fin & DEBUGCHAN_FLAG)
|
2003-03-12 06:00:18 +00:00
|
|
|
ast_frame_dump(chan->name, f, "<<");
|
2006-04-21 14:49:21 +00:00
|
|
|
chan->fin = FRAMECOUNT_INC(chan->fin);
|
2006-01-29 05:07:04 +00:00
|
|
|
|
2006-04-16 17:41:05 +00:00
|
|
|
done:
|
2007-05-22 18:52:59 +00:00
|
|
|
if (chan->music_state && chan->generator && chan->generator->digit && f && f->frametype == AST_FRAME_DTMF_END)
|
|
|
|
chan->generator->digit(chan, f->subclass);
|
|
|
|
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_unlock(chan);
|
1999-12-04 21:35:07 +00:00
|
|
|
return f;
|
|
|
|
}
|
|
|
|
|
2006-03-30 06:07:04 +00:00
|
|
|
int ast_internal_timing_enabled(struct ast_channel *chan)
|
|
|
|
{
|
2006-03-30 06:26:16 +00:00
|
|
|
int ret = ast_opt_internal_timing && chan->timingfd > -1;
|
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75983 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-19 23:24:27 +00:00
|
|
|
ast_debug(5, "Internal timing is %s (option_internal_timing=%d chan->timingfd=%d)\n", ret? "enabled": "disabled", ast_opt_internal_timing, chan->timingfd);
|
2006-03-30 06:07:04 +00:00
|
|
|
return ret;
|
|
|
|
}
|
|
|
|
|
2005-12-20 13:07:02 +00:00
|
|
|
struct ast_frame *ast_read(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
return __ast_read(chan, 0);
|
|
|
|
}
|
|
|
|
|
|
|
|
struct ast_frame *ast_read_noaudio(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
return __ast_read(chan, 1);
|
|
|
|
}
|
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
|
int ast_indicate(struct ast_channel *chan, int condition)
|
2006-05-10 12:24:11 +00:00
|
|
|
{
|
|
|
|
return ast_indicate_data(chan, condition, NULL, 0);
|
|
|
|
}
|
|
|
|
|
|
|
|
int ast_indicate_data(struct ast_channel *chan, int condition, const void *data, size_t datalen)
|
2001-05-07 00:43:32 +00:00
|
|
|
{
|
|
|
|
int res = -1;
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2006-04-19 06:58:20 +00:00
|
|
|
ast_channel_lock(chan);
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Stop if we're a zombie or need a soft hangup */
|
2006-01-24 22:38:17 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_ZOMBIE) || ast_check_hangup(chan)) {
|
2006-04-19 06:58:20 +00:00
|
|
|
ast_channel_unlock(chan);
|
2001-05-07 00:43:32 +00:00
|
|
|
return -1;
|
2006-01-24 22:38:17 +00:00
|
|
|
}
|
2005-03-04 06:47:24 +00:00
|
|
|
if (chan->tech->indicate)
|
2006-05-10 12:24:11 +00:00
|
|
|
res = chan->tech->indicate(chan, condition, data, datalen);
|
2006-04-19 06:58:20 +00:00
|
|
|
ast_channel_unlock(chan);
|
2005-03-04 06:47:24 +00:00
|
|
|
if (!chan->tech->indicate || res) {
|
2003-02-05 21:18:09 +00:00
|
|
|
/*
|
|
|
|
* Device does not support (that) indication, lets fake
|
|
|
|
* it by doing our own tone generation. (PM2002)
|
|
|
|
*/
|
2006-04-16 13:58:43 +00:00
|
|
|
if (condition < 0)
|
|
|
|
ast_playtones_stop(chan);
|
|
|
|
else {
|
2006-12-25 06:38:09 +00:00
|
|
|
const struct ind_tone_zone_sound *ts = NULL;
|
2003-02-05 21:18:09 +00:00
|
|
|
switch (condition) {
|
2005-07-25 22:13:32 +00:00
|
|
|
case AST_CONTROL_RINGING:
|
2003-02-05 21:18:09 +00:00
|
|
|
ts = ast_get_indication_tone(chan->zone, "ring");
|
|
|
|
break;
|
2005-07-25 22:13:32 +00:00
|
|
|
case AST_CONTROL_BUSY:
|
2003-02-05 21:18:09 +00:00
|
|
|
ts = ast_get_indication_tone(chan->zone, "busy");
|
|
|
|
break;
|
2005-07-25 22:13:32 +00:00
|
|
|
case AST_CONTROL_CONGESTION:
|
2003-02-05 21:18:09 +00:00
|
|
|
ts = ast_get_indication_tone(chan->zone, "congestion");
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
if (ts && ts->data[0]) {
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Driver for channel '%s' does not support indication %d, emulating it\n", chan->name, condition);
|
2003-03-16 22:37:31 +00:00
|
|
|
ast_playtones_start(chan,0,ts->data, 1);
|
2003-08-13 18:29:58 +00:00
|
|
|
res = 0;
|
2007-12-03 18:44:16 +00:00
|
|
|
chan->visible_indication = condition;
|
2003-05-15 22:39:01 +00:00
|
|
|
} else if (condition == AST_CONTROL_PROGRESS) {
|
|
|
|
/* ast_playtones_stop(chan); */
|
2004-06-14 21:18:52 +00:00
|
|
|
} else if (condition == AST_CONTROL_PROCEEDING) {
|
|
|
|
/* Do nothing, really */
|
2005-01-17 12:37:55 +00:00
|
|
|
} else if (condition == AST_CONTROL_HOLD) {
|
|
|
|
/* Do nothing.... */
|
|
|
|
} else if (condition == AST_CONTROL_UNHOLD) {
|
|
|
|
/* Do nothing.... */
|
2005-08-30 02:12:09 +00:00
|
|
|
} else if (condition == AST_CONTROL_VIDUPDATE) {
|
|
|
|
/* Do nothing.... */
|
2008-03-05 22:43:22 +00:00
|
|
|
} else if (condition == AST_CONTROL_SRCUPDATE) {
|
|
|
|
/* Do nothing... */
|
2003-05-15 22:39:01 +00:00
|
|
|
} else {
|
2003-02-05 21:18:09 +00:00
|
|
|
/* not handled */
|
|
|
|
ast_log(LOG_WARNING, "Unable to handle indication %d for '%s'\n", condition, chan->name);
|
2003-08-13 18:29:58 +00:00
|
|
|
res = -1;
|
2003-02-05 21:18:09 +00:00
|
|
|
}
|
|
|
|
}
|
2007-12-03 18:44:16 +00:00
|
|
|
} else
|
|
|
|
chan->visible_indication = condition;
|
|
|
|
|
2003-08-13 18:29:58 +00:00
|
|
|
return res;
|
2001-05-07 00:43:32 +00:00
|
|
|
}
|
|
|
|
|
2001-12-20 15:21:47 +00:00
|
|
|
int ast_recvchar(struct ast_channel *chan, int timeout)
|
|
|
|
{
|
2005-06-23 21:28:09 +00:00
|
|
|
int c;
|
|
|
|
char *buf = ast_recvtext(chan, timeout);
|
|
|
|
if (buf == NULL)
|
|
|
|
return -1; /* error or timeout */
|
|
|
|
c = *(unsigned char *)buf;
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(buf);
|
2005-06-23 21:28:09 +00:00
|
|
|
return c;
|
2001-12-20 15:21:47 +00:00
|
|
|
}
|
|
|
|
|
2005-06-21 01:16:18 +00:00
|
|
|
char *ast_recvtext(struct ast_channel *chan, int timeout)
|
|
|
|
{
|
2005-06-23 21:28:09 +00:00
|
|
|
int res, done = 0;
|
|
|
|
char *buf = NULL;
|
2005-06-21 01:16:18 +00:00
|
|
|
|
2005-06-23 21:28:09 +00:00
|
|
|
while (!done) {
|
|
|
|
struct ast_frame *f;
|
|
|
|
if (ast_check_hangup(chan))
|
|
|
|
break;
|
|
|
|
res = ast_waitfor(chan, timeout);
|
|
|
|
if (res <= 0) /* timeout or error */
|
|
|
|
break;
|
|
|
|
timeout = res; /* update timeout */
|
2005-06-21 01:16:18 +00:00
|
|
|
f = ast_read(chan);
|
2005-06-23 21:28:09 +00:00
|
|
|
if (f == NULL)
|
|
|
|
break; /* no frame */
|
|
|
|
if (f->frametype == AST_FRAME_CONTROL && f->subclass == AST_CONTROL_HANGUP)
|
|
|
|
done = 1; /* force a break */
|
2005-07-15 22:06:15 +00:00
|
|
|
else if (f->frametype == AST_FRAME_TEXT) { /* what we want */
|
2008-05-22 16:29:54 +00:00
|
|
|
buf = ast_strndup((char *) f->data.ptr, f->datalen); /* dup and break */
|
2005-06-23 21:28:09 +00:00
|
|
|
done = 1;
|
2005-06-21 01:16:18 +00:00
|
|
|
}
|
|
|
|
ast_frfree(f);
|
|
|
|
}
|
2005-06-23 21:28:09 +00:00
|
|
|
return buf;
|
2005-06-21 01:16:18 +00:00
|
|
|
}
|
|
|
|
|
2005-10-14 00:46:13 +00:00
|
|
|
int ast_sendtext(struct ast_channel *chan, const char *text)
|
2000-03-26 01:59:06 +00:00
|
|
|
{
|
|
|
|
int res = 0;
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Stop if we're a zombie or need a soft hangup */
|
2006-04-06 15:40:12 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_ZOMBIE) || ast_check_hangup(chan))
|
2001-05-07 00:43:32 +00:00
|
|
|
return -1;
|
2000-03-26 01:59:06 +00:00
|
|
|
CHECK_BLOCKING(chan);
|
2005-03-04 06:47:24 +00:00
|
|
|
if (chan->tech->send_text)
|
|
|
|
res = chan->tech->send_text(chan, text);
|
2004-12-07 20:38:43 +00:00
|
|
|
ast_clear_flag(chan, AST_FLAG_BLOCKING);
|
2000-03-26 01:59:06 +00:00
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
2006-08-31 01:59:02 +00:00
|
|
|
int ast_senddigit_begin(struct ast_channel *chan, char digit)
|
2003-02-05 21:18:09 +00:00
|
|
|
{
|
2007-05-09 19:21:35 +00:00
|
|
|
/* Device does not support DTMF tones, lets fake
|
|
|
|
* it by doing our own generation. */
|
|
|
|
static const char* dtmf_tones[] = {
|
|
|
|
"941+1336", /* 0 */
|
|
|
|
"697+1209", /* 1 */
|
|
|
|
"697+1336", /* 2 */
|
|
|
|
"697+1477", /* 3 */
|
|
|
|
"770+1209", /* 4 */
|
|
|
|
"770+1336", /* 5 */
|
|
|
|
"770+1477", /* 6 */
|
|
|
|
"852+1209", /* 7 */
|
|
|
|
"852+1336", /* 8 */
|
|
|
|
"852+1477", /* 9 */
|
|
|
|
"697+1633", /* A */
|
|
|
|
"770+1633", /* B */
|
|
|
|
"852+1633", /* C */
|
|
|
|
"941+1633", /* D */
|
|
|
|
"941+1209", /* * */
|
|
|
|
"941+1477" /* # */
|
|
|
|
};
|
2003-02-05 21:18:09 +00:00
|
|
|
|
2007-05-09 19:21:35 +00:00
|
|
|
if (!chan->tech->send_digit_begin)
|
|
|
|
return 0;
|
2006-08-31 01:59:02 +00:00
|
|
|
|
2007-05-09 19:21:35 +00:00
|
|
|
if (!chan->tech->send_digit_begin(chan, digit))
|
|
|
|
return 0;
|
|
|
|
|
|
|
|
if (digit >= '0' && digit <='9')
|
|
|
|
ast_playtones_start(chan, 0, dtmf_tones[digit-'0'], 0);
|
|
|
|
else if (digit >= 'A' && digit <= 'D')
|
|
|
|
ast_playtones_start(chan, 0, dtmf_tones[digit-'A'+10], 0);
|
|
|
|
else if (digit == '*')
|
|
|
|
ast_playtones_start(chan, 0, dtmf_tones[14], 0);
|
|
|
|
else if (digit == '#')
|
|
|
|
ast_playtones_start(chan, 0, dtmf_tones[15], 0);
|
|
|
|
else {
|
|
|
|
/* not handled */
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Unable to generate DTMF tone '%c' for '%s'\n", digit, chan->name);
|
2003-02-05 21:18:09 +00:00
|
|
|
}
|
2006-08-31 01:59:02 +00:00
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
int ast_senddigit_end(struct ast_channel *chan, char digit, unsigned int duration)
|
2006-08-31 01:59:02 +00:00
|
|
|
{
|
|
|
|
int res = -1;
|
|
|
|
|
|
|
|
if (chan->tech->send_digit_end)
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
res = chan->tech->send_digit_end(chan, digit, duration);
|
2006-08-31 01:59:02 +00:00
|
|
|
|
|
|
|
if (res && chan->generator)
|
|
|
|
ast_playtones_stop(chan);
|
|
|
|
|
2003-02-05 21:18:09 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2007-08-06 21:52:30 +00:00
|
|
|
int ast_senddigit(struct ast_channel *chan, char digit, unsigned int duration)
|
2004-05-20 00:29:09 +00:00
|
|
|
{
|
2007-05-09 16:44:33 +00:00
|
|
|
if (chan->tech->send_digit_begin) {
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
ast_senddigit_begin(chan, digit);
|
2007-08-06 21:52:30 +00:00
|
|
|
ast_safe_sleep(chan, (duration >= AST_DEFAULT_EMULATE_DTMF_DURATION ? duration : AST_DEFAULT_EMULATE_DTMF_DURATION));
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
}
|
2006-08-31 01:59:02 +00:00
|
|
|
|
2007-08-06 21:52:30 +00:00
|
|
|
return ast_senddigit_end(chan, digit, (duration >= AST_DEFAULT_EMULATE_DTMF_DURATION ? duration : AST_DEFAULT_EMULATE_DTMF_DURATION));
|
2004-05-20 00:29:09 +00:00
|
|
|
}
|
|
|
|
|
2003-04-16 02:47:03 +00:00
|
|
|
int ast_prod(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
struct ast_frame a = { AST_FRAME_VOICE };
|
|
|
|
char nothing[128];
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2003-04-16 02:47:03 +00:00
|
|
|
/* Send an empty audio frame to get things moving */
|
|
|
|
if (chan->_state != AST_STATE_UP) {
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Prodding channel '%s'\n", chan->name);
|
2005-03-04 06:47:24 +00:00
|
|
|
a.subclass = chan->rawwriteformat;
|
2008-05-22 16:29:54 +00:00
|
|
|
a.data.ptr = nothing + AST_FRIENDLY_OFFSET;
|
2004-11-01 01:44:11 +00:00
|
|
|
a.src = "ast_prod";
|
2003-04-17 02:52:53 +00:00
|
|
|
if (ast_write(chan, &a))
|
|
|
|
ast_log(LOG_WARNING, "Prodding channel '%s' failed\n", chan->name);
|
2003-04-16 02:47:03 +00:00
|
|
|
}
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2003-06-28 16:40:02 +00:00
|
|
|
int ast_write_video(struct ast_channel *chan, struct ast_frame *fr)
|
|
|
|
{
|
|
|
|
int res;
|
2005-03-04 06:47:24 +00:00
|
|
|
if (!chan->tech->write_video)
|
2003-06-28 16:40:02 +00:00
|
|
|
return 0;
|
|
|
|
res = ast_write(chan, fr);
|
|
|
|
if (!res)
|
|
|
|
res = 1;
|
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
1999-12-04 21:35:07 +00:00
|
|
|
int ast_write(struct ast_channel *chan, struct ast_frame *fr)
|
|
|
|
{
|
|
|
|
int res = -1;
|
2007-09-06 22:34:44 +00:00
|
|
|
struct ast_frame *f = NULL, *f2 = NULL;
|
2007-12-18 17:05:19 +00:00
|
|
|
int count = 0;
|
2006-04-16 13:58:43 +00:00
|
|
|
|
2007-12-18 17:05:19 +00:00
|
|
|
/*Deadlock avoidance*/
|
|
|
|
while(ast_channel_trylock(chan)) {
|
|
|
|
/*cannot goto done since the channel is not locked*/
|
|
|
|
if(count++ > 10) {
|
|
|
|
ast_debug(1, "Deadlock avoided for write to channel '%s'\n", chan->name);
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
usleep(1);
|
|
|
|
}
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Stop if we're a zombie or need a soft hangup */
|
2006-04-21 14:49:21 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_ZOMBIE) || ast_check_hangup(chan))
|
|
|
|
goto done;
|
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Handle any pending masquerades */
|
2006-04-16 13:58:43 +00:00
|
|
|
if (chan->masq && ast_do_masquerade(chan)) {
|
|
|
|
ast_log(LOG_WARNING, "Failed to perform masquerade\n");
|
2006-04-21 14:49:21 +00:00
|
|
|
goto done;
|
2001-05-07 00:43:32 +00:00
|
|
|
}
|
2003-07-29 16:09:03 +00:00
|
|
|
if (chan->masqr) {
|
2006-04-21 14:49:21 +00:00
|
|
|
res = 0; /* XXX explain, why 0 ? */
|
|
|
|
goto done;
|
2003-07-29 16:09:03 +00:00
|
|
|
}
|
2002-06-21 01:40:13 +00:00
|
|
|
if (chan->generatordata) {
|
2004-12-07 20:38:43 +00:00
|
|
|
if (ast_test_flag(chan, AST_FLAG_WRITE_INT))
|
2004-04-07 14:43:18 +00:00
|
|
|
ast_deactivate_generator(chan);
|
2003-07-29 16:09:03 +00:00
|
|
|
else {
|
2007-05-02 23:00:07 +00:00
|
|
|
if (fr->frametype == AST_FRAME_DTMF_END) {
|
|
|
|
/* There is a generator running while we're in the middle of a digit.
|
|
|
|
* It's probably inband DTMF, so go ahead and pass it so it can
|
|
|
|
* stop the generator */
|
|
|
|
ast_clear_flag(chan, AST_FLAG_BLOCKING);
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
res = ast_senddigit_end(chan, fr->subclass, fr->len);
|
|
|
|
ast_channel_lock(chan);
|
|
|
|
CHECK_BLOCKING(chan);
|
2007-05-14 17:25:25 +00:00
|
|
|
} else if (fr->frametype == AST_FRAME_CONTROL && fr->subclass == AST_CONTROL_UNHOLD) {
|
|
|
|
/* This is a side case where Echo is basically being called and the person put themselves on hold and took themselves off hold */
|
|
|
|
res = (chan->tech->indicate == NULL) ? 0 :
|
2008-05-22 16:29:54 +00:00
|
|
|
chan->tech->indicate(chan, fr->subclass, fr->data.ptr, fr->datalen);
|
2007-05-02 23:00:07 +00:00
|
|
|
}
|
2006-04-21 14:49:21 +00:00
|
|
|
res = 0; /* XXX explain, why 0 ? */
|
|
|
|
goto done;
|
2003-07-29 16:09:03 +00:00
|
|
|
}
|
2002-06-21 01:40:13 +00:00
|
|
|
}
|
2005-05-14 23:26:37 +00:00
|
|
|
/* High bit prints debugging */
|
2006-04-21 14:49:21 +00:00
|
|
|
if (chan->fout & DEBUGCHAN_FLAG)
|
2003-03-12 06:00:18 +00:00
|
|
|
ast_frame_dump(chan->name, fr, ">>");
|
1999-12-04 21:35:07 +00:00
|
|
|
CHECK_BLOCKING(chan);
|
2007-01-23 00:11:32 +00:00
|
|
|
switch (fr->frametype) {
|
1999-12-04 21:35:07 +00:00
|
|
|
case AST_FRAME_CONTROL:
|
2006-07-23 05:06:47 +00:00
|
|
|
res = (chan->tech->indicate == NULL) ? 0 :
|
2008-05-22 16:29:54 +00:00
|
|
|
chan->tech->indicate(chan, fr->subclass, fr->data.ptr, fr->datalen);
|
1999-12-04 21:35:07 +00:00
|
|
|
break;
|
2006-01-29 05:07:04 +00:00
|
|
|
case AST_FRAME_DTMF_BEGIN:
|
2007-09-06 22:34:44 +00:00
|
|
|
if (chan->audiohooks) {
|
|
|
|
struct ast_frame *old_frame = fr;
|
2007-08-08 19:30:52 +00:00
|
|
|
fr = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_WRITE, fr);
|
2007-09-06 22:34:44 +00:00
|
|
|
if (old_frame != fr)
|
|
|
|
f = fr;
|
|
|
|
}
|
2007-04-10 23:55:26 +00:00
|
|
|
send_dtmf_event(chan, "Sent", fr->subclass, "Yes", "No");
|
2006-08-31 01:59:02 +00:00
|
|
|
ast_clear_flag(chan, AST_FLAG_BLOCKING);
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
res = ast_senddigit_begin(chan, fr->subclass);
|
|
|
|
ast_channel_lock(chan);
|
|
|
|
CHECK_BLOCKING(chan);
|
2006-01-29 05:15:24 +00:00
|
|
|
break;
|
2006-01-29 05:07:04 +00:00
|
|
|
case AST_FRAME_DTMF_END:
|
2007-09-06 22:34:44 +00:00
|
|
|
if (chan->audiohooks) {
|
|
|
|
struct ast_frame *old_frame = fr;
|
2007-08-08 19:30:52 +00:00
|
|
|
fr = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_WRITE, fr);
|
2007-09-06 22:34:44 +00:00
|
|
|
if (old_frame != fr)
|
|
|
|
f = fr;
|
|
|
|
}
|
2007-04-10 23:55:26 +00:00
|
|
|
send_dtmf_event(chan, "Sent", fr->subclass, "No", "Yes");
|
2004-12-07 20:38:43 +00:00
|
|
|
ast_clear_flag(chan, AST_FLAG_BLOCKING);
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_unlock(chan);
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
res = ast_senddigit_end(chan, fr->subclass, fr->len);
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_lock(chan);
|
2003-09-24 19:43:14 +00:00
|
|
|
CHECK_BLOCKING(chan);
|
1999-12-04 21:35:07 +00:00
|
|
|
break;
|
2001-12-20 15:21:47 +00:00
|
|
|
case AST_FRAME_TEXT:
|
2007-02-16 13:35:44 +00:00
|
|
|
if (fr->subclass == AST_FORMAT_T140) {
|
|
|
|
res = (chan->tech->write_text == NULL) ? 0 :
|
|
|
|
chan->tech->write_text(chan, fr);
|
|
|
|
} else {
|
|
|
|
res = (chan->tech->send_text == NULL) ? 0 :
|
2008-05-22 16:29:54 +00:00
|
|
|
chan->tech->send_text(chan, (char *) fr->data.ptr);
|
2007-02-16 13:35:44 +00:00
|
|
|
}
|
2005-02-28 06:06:42 +00:00
|
|
|
break;
|
|
|
|
case AST_FRAME_HTML:
|
2006-04-16 13:58:43 +00:00
|
|
|
res = (chan->tech->send_html == NULL) ? 0 :
|
2008-05-22 16:29:54 +00:00
|
|
|
chan->tech->send_html(chan, fr->subclass, (char *) fr->data.ptr, fr->datalen);
|
2001-12-20 15:21:47 +00:00
|
|
|
break;
|
2003-06-28 16:40:02 +00:00
|
|
|
case AST_FRAME_VIDEO:
|
|
|
|
/* XXX Handle translation of video codecs one day XXX */
|
2006-04-16 13:58:43 +00:00
|
|
|
res = (chan->tech->write_video == NULL) ? 0 :
|
|
|
|
chan->tech->write_video(chan, fr);
|
2003-06-28 16:40:02 +00:00
|
|
|
break;
|
2006-06-13 18:35:53 +00:00
|
|
|
case AST_FRAME_MODEM:
|
|
|
|
res = (chan->tech->write == NULL) ? 0 :
|
|
|
|
chan->tech->write(chan, fr);
|
|
|
|
break;
|
2006-01-29 05:07:04 +00:00
|
|
|
case AST_FRAME_VOICE:
|
2006-04-16 13:58:43 +00:00
|
|
|
if (chan->tech->write == NULL)
|
2006-04-19 06:58:20 +00:00
|
|
|
break; /*! \todo XXX should return 0 maybe ? */
|
2006-04-16 13:58:43 +00:00
|
|
|
|
2007-08-08 19:30:52 +00:00
|
|
|
/* If audiohooks are present, write the frame out */
|
2007-09-06 22:34:44 +00:00
|
|
|
if (chan->audiohooks) {
|
|
|
|
struct ast_frame *old_frame = fr;
|
2007-08-08 19:30:52 +00:00
|
|
|
fr = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_WRITE, fr);
|
2007-09-06 22:34:44 +00:00
|
|
|
if (old_frame != fr)
|
|
|
|
f2 = fr;
|
|
|
|
}
|
|
|
|
|
2007-08-08 19:30:52 +00:00
|
|
|
/* If the frame is in the raw write format, then it's easy... just use the frame - otherwise we will have to translate */
|
|
|
|
if (fr->subclass == chan->rawwriteformat)
|
|
|
|
f = fr;
|
|
|
|
else
|
|
|
|
f = (chan->writetrans) ? ast_translate(chan->writetrans, fr, 0) : fr;
|
2006-09-26 20:11:44 +00:00
|
|
|
|
2007-08-08 19:30:52 +00:00
|
|
|
if (!f) {
|
2006-04-16 14:14:16 +00:00
|
|
|
res = 0;
|
2006-09-26 20:11:44 +00:00
|
|
|
break;
|
|
|
|
}
|
2005-03-24 03:04:45 +00:00
|
|
|
|
2006-09-26 20:11:44 +00:00
|
|
|
/* If Monitor is running on this channel, then we have to write frames out there too */
|
|
|
|
if (chan->monitor && chan->monitor->write_stream) {
|
|
|
|
/* XXX must explain this code */
|
2003-08-28 20:02:10 +00:00
|
|
|
#ifndef MONITOR_CONSTANT_DELAY
|
2006-09-26 20:11:44 +00:00
|
|
|
int jump = chan->insmpl - chan->outsmpl - 4 * f->samples;
|
|
|
|
if (jump >= 0) {
|
2007-01-24 01:00:57 +00:00
|
|
|
jump = chan->insmpl - chan->outsmpl;
|
|
|
|
if (ast_seekstream(chan->monitor->write_stream, jump, SEEK_FORCECUR) == -1)
|
2006-09-26 20:11:44 +00:00
|
|
|
ast_log(LOG_WARNING, "Failed to perform seek in monitoring write stream, synchronization between the files may be broken\n");
|
2007-01-24 01:00:57 +00:00
|
|
|
chan->outsmpl += jump + f->samples;
|
2006-09-26 20:11:44 +00:00
|
|
|
} else
|
|
|
|
chan->outsmpl += f->samples;
|
2003-08-28 20:02:10 +00:00
|
|
|
#else
|
2006-09-26 20:11:44 +00:00
|
|
|
int jump = chan->insmpl - chan->outsmpl;
|
|
|
|
if (jump - MONITOR_DELAY >= 0) {
|
|
|
|
if (ast_seekstream(chan->monitor->write_stream, jump - f->samples, SEEK_FORCECUR) == -1)
|
|
|
|
ast_log(LOG_WARNING, "Failed to perform seek in monitoring write stream, synchronization between the files may be broken\n");
|
|
|
|
chan->outsmpl += jump;
|
|
|
|
} else
|
|
|
|
chan->outsmpl += f->samples;
|
2003-08-28 20:02:10 +00:00
|
|
|
#endif
|
2006-09-26 20:11:44 +00:00
|
|
|
if (chan->monitor->state == AST_MONITOR_RUNNING) {
|
|
|
|
if (ast_writestream(chan->monitor->write_stream, f) < 0)
|
|
|
|
ast_log(LOG_WARNING, "Failed to write data to channel monitor write stream\n");
|
2006-04-16 13:58:43 +00:00
|
|
|
}
|
2006-09-26 20:11:44 +00:00
|
|
|
}
|
2007-10-09 22:21:49 +00:00
|
|
|
|
2007-12-18 17:05:19 +00:00
|
|
|
if (f)
|
|
|
|
res = chan->tech->write(chan,f);
|
|
|
|
else
|
2007-10-09 22:21:49 +00:00
|
|
|
res = 0;
|
2006-08-31 01:59:02 +00:00
|
|
|
break;
|
|
|
|
case AST_FRAME_NULL:
|
|
|
|
case AST_FRAME_IAX:
|
|
|
|
/* Ignore these */
|
2006-08-31 18:47:51 +00:00
|
|
|
res = 0;
|
2006-08-31 01:59:02 +00:00
|
|
|
break;
|
|
|
|
default:
|
2006-10-24 03:09:48 +00:00
|
|
|
/* At this point, fr is the incoming frame and f is NULL. Channels do
|
|
|
|
* not expect to get NULL as a frame pointer and will segfault. Hence,
|
|
|
|
* we output the original frame passed in. */
|
|
|
|
res = chan->tech->write(chan, fr);
|
2006-08-31 01:59:02 +00:00
|
|
|
break;
|
1999-12-04 21:35:07 +00:00
|
|
|
}
|
2005-08-22 21:19:59 +00:00
|
|
|
|
2006-04-16 13:58:43 +00:00
|
|
|
if (f && f != fr)
|
2003-02-12 13:59:15 +00:00
|
|
|
ast_frfree(f);
|
2007-09-06 22:34:44 +00:00
|
|
|
if (f2)
|
|
|
|
ast_frfree(f2);
|
2004-12-07 20:38:43 +00:00
|
|
|
ast_clear_flag(chan, AST_FLAG_BLOCKING);
|
2002-06-21 01:40:13 +00:00
|
|
|
/* Consider a write failure to force a soft hangup */
|
|
|
|
if (res < 0)
|
2002-09-10 20:48:20 +00:00
|
|
|
chan->_softhangup |= AST_SOFTHANGUP_DEV;
|
2003-03-12 06:00:18 +00:00
|
|
|
else {
|
2006-04-21 14:49:21 +00:00
|
|
|
chan->fout = FRAMECOUNT_INC(chan->fout);
|
2003-03-12 06:00:18 +00:00
|
|
|
}
|
2006-04-21 14:49:21 +00:00
|
|
|
done:
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_unlock(chan);
|
1999-12-04 21:35:07 +00:00
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
2005-04-04 03:28:38 +00:00
|
|
|
static int set_format(struct ast_channel *chan, int fmt, int *rawformat, int *format,
|
|
|
|
struct ast_trans_pvt **trans, const int direction)
|
2001-03-12 03:04:51 +00:00
|
|
|
{
|
|
|
|
int native;
|
|
|
|
int res;
|
2008-06-19 17:02:54 +00:00
|
|
|
char from[200], to[200];
|
|
|
|
|
2008-02-19 22:35:23 +00:00
|
|
|
/* Make sure we only consider audio */
|
|
|
|
fmt &= AST_FORMAT_AUDIO_MASK;
|
|
|
|
|
2008-02-19 22:23:04 +00:00
|
|
|
native = chan->nativeformats;
|
|
|
|
|
2008-02-18 17:12:13 +00:00
|
|
|
if (!fmt || !native) /* No audio requested */
|
|
|
|
return 0; /* Let's try a call without any sounds (video, text) */
|
2001-03-12 03:04:51 +00:00
|
|
|
|
2005-04-04 03:28:38 +00:00
|
|
|
/* Find a translation path from the native format to one of the desired formats */
|
|
|
|
if (!direction)
|
|
|
|
/* reading */
|
|
|
|
res = ast_translator_best_choice(&fmt, &native);
|
|
|
|
else
|
|
|
|
/* writing */
|
|
|
|
res = ast_translator_best_choice(&native, &fmt);
|
|
|
|
|
2001-03-12 03:04:51 +00:00
|
|
|
if (res < 0) {
|
2005-07-25 22:13:32 +00:00
|
|
|
ast_log(LOG_WARNING, "Unable to find a codec translation path from %s to %s\n",
|
2008-06-19 17:02:54 +00:00
|
|
|
ast_getformatname_multiple(from, sizeof(from), native),
|
|
|
|
ast_getformatname_multiple(to, sizeof(to), fmt));
|
2001-03-12 03:04:51 +00:00
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
2005-04-04 03:28:38 +00:00
|
|
|
/* Now we have a good choice for both. */
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_lock(chan);
|
2006-07-27 15:42:49 +00:00
|
|
|
|
2006-09-09 20:25:45 +00:00
|
|
|
if ((*rawformat == native) && (*format == fmt) && ((*rawformat == *format) || (*trans))) {
|
2006-07-27 15:42:49 +00:00
|
|
|
/* the channel is already in these formats, so nothing to do */
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2005-04-04 03:28:38 +00:00
|
|
|
*rawformat = native;
|
2001-03-12 03:04:51 +00:00
|
|
|
/* User perspective is fmt */
|
2005-04-04 03:28:38 +00:00
|
|
|
*format = fmt;
|
|
|
|
/* Free any read translation we have right now */
|
|
|
|
if (*trans)
|
|
|
|
ast_translator_free_path(*trans);
|
|
|
|
/* Build a translation path from the raw format to the desired format */
|
|
|
|
if (!direction)
|
|
|
|
/* reading */
|
|
|
|
*trans = ast_translator_build_path(*format, *rawformat);
|
|
|
|
else
|
|
|
|
/* writing */
|
|
|
|
*trans = ast_translator_build_path(*rawformat, *format);
|
2006-04-15 15:07:39 +00:00
|
|
|
ast_channel_unlock(chan);
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Set channel %s to %s format %s\n", chan->name,
|
|
|
|
direction ? "write" : "read", ast_getformatname(fmt));
|
2001-03-12 03:04:51 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2005-04-04 03:28:38 +00:00
|
|
|
int ast_set_read_format(struct ast_channel *chan, int fmt)
|
2001-03-12 03:04:51 +00:00
|
|
|
{
|
2005-04-04 03:28:38 +00:00
|
|
|
return set_format(chan, fmt, &chan->rawreadformat, &chan->readformat,
|
|
|
|
&chan->readtrans, 0);
|
|
|
|
}
|
|
|
|
|
|
|
|
int ast_set_write_format(struct ast_channel *chan, int fmt)
|
|
|
|
{
|
|
|
|
return set_format(chan, fmt, &chan->rawwriteformat, &chan->writeformat,
|
|
|
|
&chan->writetrans, 1);
|
2001-03-12 03:04:51 +00:00
|
|
|
}
|
|
|
|
|
2007-08-13 14:23:38 +00:00
|
|
|
const char *ast_channel_reason2str(int reason)
|
2007-08-10 21:03:06 +00:00
|
|
|
{
|
|
|
|
switch (reason) /* the following appear to be the only ones actually returned by request_and_dial */
|
|
|
|
{
|
|
|
|
case 0:
|
|
|
|
return "Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?)";
|
|
|
|
case AST_CONTROL_HANGUP:
|
|
|
|
return "Hangup";
|
|
|
|
case AST_CONTROL_RING:
|
|
|
|
return "Local Ring";
|
|
|
|
case AST_CONTROL_RINGING:
|
|
|
|
return "Remote end Ringing";
|
|
|
|
case AST_CONTROL_ANSWER:
|
|
|
|
return "Remote end has Answered";
|
|
|
|
case AST_CONTROL_BUSY:
|
|
|
|
return "Remote end is Busy";
|
|
|
|
case AST_CONTROL_CONGESTION:
|
|
|
|
return "Congestion (circuits busy)";
|
|
|
|
default:
|
|
|
|
return "Unknown Reason!!";
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2004-10-03 04:19:59 +00:00
|
|
|
struct ast_channel *__ast_request_and_dial(const char *type, int format, void *data, int timeout, int *outstate, const char *cid_num, const char *cid_name, struct outgoing_helper *oh)
|
2002-09-10 20:48:20 +00:00
|
|
|
{
|
2006-04-16 15:13:39 +00:00
|
|
|
int dummy_outstate;
|
2004-10-26 22:25:43 +00:00
|
|
|
int cause = 0;
|
2002-09-10 20:48:20 +00:00
|
|
|
struct ast_channel *chan;
|
2003-09-12 16:51:35 +00:00
|
|
|
int res = 0;
|
2008-01-11 19:12:05 +00:00
|
|
|
int last_subclass = 0;
|
2005-07-18 05:00:23 +00:00
|
|
|
|
2006-04-16 15:13:39 +00:00
|
|
|
if (outstate)
|
|
|
|
*outstate = 0;
|
|
|
|
else
|
|
|
|
outstate = &dummy_outstate; /* make outstate always a valid pointer */
|
|
|
|
|
2004-10-26 22:25:43 +00:00
|
|
|
chan = ast_request(type, format, data, &cause);
|
2006-04-16 15:13:39 +00:00
|
|
|
if (!chan) {
|
|
|
|
ast_log(LOG_NOTICE, "Unable to request channel %s/%s\n", type, (char *)data);
|
|
|
|
/* compute error and return */
|
|
|
|
if (cause == AST_CAUSE_BUSY)
|
|
|
|
*outstate = AST_CONTROL_BUSY;
|
|
|
|
else if (cause == AST_CAUSE_CONGESTION)
|
|
|
|
*outstate = AST_CONTROL_CONGESTION;
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (oh) {
|
|
|
|
if (oh->vars)
|
|
|
|
ast_set_variables(chan, oh->vars);
|
|
|
|
/* XXX why is this necessary, for the parent_channel perhaps ? */
|
|
|
|
if (!ast_strlen_zero(oh->cid_num) && !ast_strlen_zero(oh->cid_name))
|
|
|
|
ast_set_callerid(chan, oh->cid_num, oh->cid_name, oh->cid_num);
|
2007-03-03 14:40:18 +00:00
|
|
|
if (oh->parent_channel) {
|
2006-04-16 15:13:39 +00:00
|
|
|
ast_channel_inherit_variables(oh->parent_channel, chan);
|
2007-03-03 14:40:18 +00:00
|
|
|
ast_channel_datastore_inherit(oh->parent_channel, chan);
|
|
|
|
}
|
2006-04-16 15:13:39 +00:00
|
|
|
if (oh->account)
|
|
|
|
ast_cdr_setaccount(chan, oh->account);
|
|
|
|
}
|
|
|
|
ast_set_callerid(chan, cid_num, cid_name, cid_num);
|
|
|
|
|
|
|
|
if (ast_call(chan, data, 0)) { /* ast_call failed... */
|
|
|
|
ast_log(LOG_NOTICE, "Unable to call channel %s/%s\n", type, (char *)data);
|
|
|
|
} else {
|
|
|
|
res = 1; /* mark success in case chan->_state is already AST_STATE_UP */
|
|
|
|
while (timeout && chan->_state != AST_STATE_UP) {
|
|
|
|
struct ast_frame *f;
|
|
|
|
res = ast_waitfor(chan, timeout);
|
|
|
|
if (res <= 0) /* error, timeout, or done */
|
|
|
|
break;
|
|
|
|
if (timeout > -1)
|
|
|
|
timeout = res;
|
|
|
|
f = ast_read(chan);
|
|
|
|
if (!f) {
|
|
|
|
*outstate = AST_CONTROL_HANGUP;
|
|
|
|
res = 0;
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
if (f->frametype == AST_FRAME_CONTROL) {
|
|
|
|
switch (f->subclass) {
|
|
|
|
case AST_CONTROL_RINGING: /* record but keep going */
|
|
|
|
*outstate = f->subclass;
|
2002-09-10 20:48:20 +00:00
|
|
|
break;
|
2006-04-16 15:13:39 +00:00
|
|
|
|
|
|
|
case AST_CONTROL_BUSY:
|
|
|
|
case AST_CONTROL_CONGESTION:
|
|
|
|
case AST_CONTROL_ANSWER:
|
|
|
|
*outstate = f->subclass;
|
|
|
|
timeout = 0; /* trick to force exit from the while() */
|
2002-09-10 20:48:20 +00:00
|
|
|
break;
|
2006-04-16 15:13:39 +00:00
|
|
|
|
2006-05-31 16:04:49 +00:00
|
|
|
/* Ignore these */
|
|
|
|
case AST_CONTROL_PROGRESS:
|
|
|
|
case AST_CONTROL_PROCEEDING:
|
|
|
|
case AST_CONTROL_HOLD:
|
|
|
|
case AST_CONTROL_UNHOLD:
|
|
|
|
case AST_CONTROL_VIDUPDATE:
|
2008-03-05 22:43:22 +00:00
|
|
|
case AST_CONTROL_SRCUPDATE:
|
2006-04-16 15:13:39 +00:00
|
|
|
case -1: /* Ignore -- just stopping indications */
|
2002-09-10 20:48:20 +00:00
|
|
|
break;
|
2006-04-16 15:13:39 +00:00
|
|
|
|
|
|
|
default:
|
|
|
|
ast_log(LOG_NOTICE, "Don't know what to do with control frame %d\n", f->subclass);
|
2002-09-10 20:48:20 +00:00
|
|
|
}
|
2008-01-11 19:12:05 +00:00
|
|
|
last_subclass = f->subclass;
|
2002-09-10 20:48:20 +00:00
|
|
|
}
|
2006-04-16 15:13:39 +00:00
|
|
|
ast_frfree(f);
|
2004-10-26 22:25:43 +00:00
|
|
|
}
|
|
|
|
}
|
2006-04-16 15:13:39 +00:00
|
|
|
|
|
|
|
/* Final fixups */
|
|
|
|
if (oh) {
|
|
|
|
if (!ast_strlen_zero(oh->context))
|
|
|
|
ast_copy_string(chan->context, oh->context, sizeof(chan->context));
|
|
|
|
if (!ast_strlen_zero(oh->exten))
|
|
|
|
ast_copy_string(chan->exten, oh->exten, sizeof(chan->exten));
|
|
|
|
if (oh->priority)
|
|
|
|
chan->priority = oh->priority;
|
2003-10-09 14:12:26 +00:00
|
|
|
}
|
2006-04-16 15:13:39 +00:00
|
|
|
if (chan->_state == AST_STATE_UP)
|
|
|
|
*outstate = AST_CONTROL_ANSWER;
|
|
|
|
|
|
|
|
if (res <= 0) {
|
2008-01-11 19:12:05 +00:00
|
|
|
if ( AST_CONTROL_RINGING == last_subclass )
|
|
|
|
chan->hangupcause = AST_CAUSE_NO_ANSWER;
|
2006-04-16 15:13:39 +00:00
|
|
|
if (!chan->cdr && (chan->cdr = ast_cdr_alloc()))
|
2006-02-15 01:48:54 +00:00
|
|
|
ast_cdr_init(chan->cdr, chan);
|
2003-09-12 16:51:35 +00:00
|
|
|
if (chan->cdr) {
|
|
|
|
char tmp[256];
|
2006-04-16 15:13:39 +00:00
|
|
|
snprintf(tmp, sizeof(tmp), "%s/%s", type, (char *)data);
|
2003-09-12 16:51:35 +00:00
|
|
|
ast_cdr_setapp(chan->cdr,"Dial",tmp);
|
|
|
|
ast_cdr_update(chan);
|
|
|
|
ast_cdr_start(chan->cdr);
|
|
|
|
ast_cdr_end(chan->cdr);
|
|
|
|
/* If the cause wasn't handled properly */
|
|
|
|
if (ast_cdr_disposition(chan->cdr,chan->hangupcause))
|
|
|
|
ast_cdr_failed(chan->cdr);
|
2006-02-15 01:48:54 +00:00
|
|
|
}
|
2003-09-12 16:51:35 +00:00
|
|
|
ast_hangup(chan);
|
|
|
|
chan = NULL;
|
|
|
|
}
|
2002-09-10 20:48:20 +00:00
|
|
|
return chan;
|
|
|
|
}
|
|
|
|
|
2004-10-03 04:19:59 +00:00
|
|
|
struct ast_channel *ast_request_and_dial(const char *type, int format, void *data, int timeout, int *outstate, const char *cidnum, const char *cidname)
|
2003-09-22 15:27:09 +00:00
|
|
|
{
|
2004-10-02 00:58:31 +00:00
|
|
|
return __ast_request_and_dial(type, format, data, timeout, outstate, cidnum, cidname, NULL);
|
2003-09-22 15:27:09 +00:00
|
|
|
}
|
|
|
|
|
2004-10-26 22:25:43 +00:00
|
|
|
struct ast_channel *ast_request(const char *type, int format, void *data, int *cause)
|
1999-12-04 21:35:07 +00:00
|
|
|
{
|
|
|
|
struct chanlist *chan;
|
2005-09-29 17:40:24 +00:00
|
|
|
struct ast_channel *c;
|
2001-03-12 03:04:51 +00:00
|
|
|
int capabilities;
|
|
|
|
int fmt;
|
|
|
|
int res;
|
2004-10-26 22:25:43 +00:00
|
|
|
int foo;
|
2006-06-06 16:09:33 +00:00
|
|
|
int videoformat = format & AST_FORMAT_VIDEO_MASK;
|
2007-02-16 13:35:44 +00:00
|
|
|
int textformat = format & AST_FORMAT_TEXT_MASK;
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2004-10-26 22:25:43 +00:00
|
|
|
if (!cause)
|
|
|
|
cause = &foo;
|
|
|
|
*cause = AST_CAUSE_NOTDEFINED;
|
2005-09-29 17:40:24 +00:00
|
|
|
|
2007-06-11 14:41:39 +00:00
|
|
|
if (AST_RWLIST_RDLOCK(&channels)) {
|
1999-12-04 21:35:07 +00:00
|
|
|
ast_log(LOG_WARNING, "Unable to lock channel list\n");
|
|
|
|
return NULL;
|
|
|
|
}
|
2005-09-29 17:40:24 +00:00
|
|
|
|
2006-01-11 05:05:13 +00:00
|
|
|
AST_LIST_TRAVERSE(&backends, chan, list) {
|
2005-09-29 17:40:24 +00:00
|
|
|
if (strcasecmp(type, chan->tech->type))
|
|
|
|
continue;
|
|
|
|
|
|
|
|
capabilities = chan->tech->capabilities;
|
2006-06-06 16:09:33 +00:00
|
|
|
fmt = format & AST_FORMAT_AUDIO_MASK;
|
2008-02-18 17:12:13 +00:00
|
|
|
if (fmt) {
|
|
|
|
/* We have audio - is it possible to connect the various calls to each other?
|
|
|
|
(Avoid this check for calls without audio, like text+video calls)
|
|
|
|
*/
|
|
|
|
res = ast_translator_best_choice(&fmt, &capabilities);
|
|
|
|
if (res < 0) {
|
|
|
|
ast_log(LOG_WARNING, "No translator path exists for channel type %s (native 0x%x) to 0x%x\n", type, chan->tech->capabilities, format);
|
|
|
|
*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
|
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
|
|
|
return NULL;
|
|
|
|
}
|
1999-12-04 21:35:07 +00:00
|
|
|
}
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
2005-09-29 17:40:24 +00:00
|
|
|
if (!chan->tech->requester)
|
|
|
|
return NULL;
|
|
|
|
|
2007-02-16 13:35:44 +00:00
|
|
|
if (!(c = chan->tech->requester(type, capabilities | videoformat | textformat, data, cause)))
|
2005-09-29 17:40:24 +00:00
|
|
|
return NULL;
|
2006-11-07 21:47:49 +00:00
|
|
|
|
|
|
|
/* no need to generate a Newchannel event here; it is done in the channel_alloc call */
|
2005-09-29 17:40:24 +00:00
|
|
|
return c;
|
2004-10-26 22:25:43 +00:00
|
|
|
}
|
2005-09-29 17:40:24 +00:00
|
|
|
|
|
|
|
ast_log(LOG_WARNING, "No channel type registered for '%s'\n", type);
|
|
|
|
*cause = AST_CAUSE_NOSUCHDRIVER;
|
2007-06-11 14:41:39 +00:00
|
|
|
AST_RWLIST_UNLOCK(&channels);
|
2005-09-29 17:40:24 +00:00
|
|
|
|
|
|
|
return NULL;
|
1999-12-04 21:35:07 +00:00
|
|
|
}
|
|
|
|
|
2006-04-06 15:40:12 +00:00
|
|
|
int ast_call(struct ast_channel *chan, char *addr, int timeout)
|
1999-12-04 21:35:07 +00:00
|
|
|
{
|
2006-04-06 15:40:12 +00:00
|
|
|
/* Place an outgoing call, but don't wait any longer than timeout ms before returning.
|
|
|
|
If the remote end does not answer within the timeout, then do NOT hang up, but
|
1999-12-04 21:35:07 +00:00
|
|
|
return anyway. */
|
|
|
|
int res = -1;
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Stop if we're a zombie or need a soft hangup */
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_lock(chan);
|
2006-05-09 15:01:10 +00:00
|
|
|
if (!ast_test_flag(chan, AST_FLAG_ZOMBIE) && !ast_check_hangup(chan)) {
|
Merged revisions 135799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines
(closes issue #12982)
Reported by: bcnit
Tested by: murf
I discovered that also, in the previous bug fixes and changes,
the cdr.conf 'unanswered' option is not being obeyed, so
I fixed this.
And, yes, there are two 'answer' times involved in this
scenario, and I would agree with you, that the first
answer time is the time that should appear in the CDR.
(the second 'answer' time is the time that the bridge
was begun).
I made the necessary adjustments, recording the first
answer time into the peer cdr, and then using that to
override the bridge cdr's value.
To get the 'unanswered' CDRs to appear, I purposely
output them, using the dial cmd to mark them as
DIALED (with a new flag), and outputting them if
they bear that flag, and you are in the right mode.
I also corrected one small mention of the Zap device
to equally consider the dahdi device.
I heavily tested 10-sec-wait macros in dial, and
without the macro call; I tested hangups while the
macro was running vs. letting the macro complete
and the bridge form. Looks OK. Removed all the
instrumentation and debug.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135821 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05 23:45:32 +00:00
|
|
|
if (chan->cdr)
|
|
|
|
ast_set_flag(chan->cdr, AST_CDR_FLAG_DIALED);
|
2005-03-04 06:47:24 +00:00
|
|
|
if (chan->tech->call)
|
|
|
|
res = chan->tech->call(chan, addr, timeout);
|
2006-05-09 15:01:10 +00:00
|
|
|
ast_set_flag(chan, AST_FLAG_OUTGOING);
|
|
|
|
}
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_unlock(chan);
|
1999-12-04 21:35:07 +00:00
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
2006-04-06 15:40:12 +00:00
|
|
|
/*!
|
2005-12-31 16:35:48 +00:00
|
|
|
\brief Transfer a call to dest, if the channel supports transfer
|
|
|
|
|
2006-04-06 15:40:12 +00:00
|
|
|
Called by:
|
2008-03-04 23:04:29 +00:00
|
|
|
\arg app_transfer
|
|
|
|
\arg the manager interface
|
2005-12-31 16:35:48 +00:00
|
|
|
*/
|
2006-04-06 15:40:12 +00:00
|
|
|
int ast_transfer(struct ast_channel *chan, char *dest)
|
2003-05-14 05:33:06 +00:00
|
|
|
{
|
|
|
|
int res = -1;
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2003-05-14 05:33:06 +00:00
|
|
|
/* Stop if we're a zombie or need a soft hangup */
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_lock(chan);
|
2004-12-07 20:38:43 +00:00
|
|
|
if (!ast_test_flag(chan, AST_FLAG_ZOMBIE) && !ast_check_hangup(chan)) {
|
2005-03-04 06:47:24 +00:00
|
|
|
if (chan->tech->transfer) {
|
|
|
|
res = chan->tech->transfer(chan, dest);
|
2003-05-14 05:33:06 +00:00
|
|
|
if (!res)
|
|
|
|
res = 1;
|
|
|
|
} else
|
|
|
|
res = 0;
|
|
|
|
}
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_unlock(chan);
|
2003-05-14 05:33:06 +00:00
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
1999-12-04 21:35:07 +00:00
|
|
|
int ast_readstring(struct ast_channel *c, char *s, int len, int timeout, int ftimeout, char *enders)
|
|
|
|
{
|
2006-04-16 07:53:25 +00:00
|
|
|
return ast_readstring_full(c, s, len, timeout, ftimeout, enders, -1, -1);
|
2003-02-23 06:00:11 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
int ast_readstring_full(struct ast_channel *c, char *s, int len, int timeout, int ftimeout, char *enders, int audiofd, int ctrlfd)
|
|
|
|
{
|
2006-04-16 08:01:10 +00:00
|
|
|
int pos = 0; /* index in the buffer where we accumulate digits */
|
2003-02-23 06:00:11 +00:00
|
|
|
int to = ftimeout;
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2003-02-23 06:00:11 +00:00
|
|
|
/* Stop if we're a zombie or need a soft hangup */
|
2006-04-06 15:40:12 +00:00
|
|
|
if (ast_test_flag(c, AST_FLAG_ZOMBIE) || ast_check_hangup(c))
|
2003-02-23 06:00:11 +00:00
|
|
|
return -1;
|
|
|
|
if (!len)
|
|
|
|
return -1;
|
2006-04-14 20:47:06 +00:00
|
|
|
for (;;) {
|
2006-04-16 08:01:10 +00:00
|
|
|
int d;
|
2003-06-29 21:54:58 +00:00
|
|
|
if (c->stream) {
|
2003-02-23 06:00:11 +00:00
|
|
|
d = ast_waitstream_full(c, AST_DIGIT_ANY, audiofd, ctrlfd);
|
|
|
|
ast_stopstream(c);
|
|
|
|
usleep(1000);
|
|
|
|
if (!d)
|
|
|
|
d = ast_waitfordigit_full(c, to, audiofd, ctrlfd);
|
|
|
|
} else {
|
|
|
|
d = ast_waitfordigit_full(c, to, audiofd, ctrlfd);
|
|
|
|
}
|
|
|
|
if (d < 0)
|
|
|
|
return -1;
|
|
|
|
if (d == 0) {
|
|
|
|
s[pos]='\0';
|
|
|
|
return 1;
|
|
|
|
}
|
|
|
|
if (d == 1) {
|
|
|
|
s[pos]='\0';
|
|
|
|
return 2;
|
|
|
|
}
|
|
|
|
if (!strchr(enders, d))
|
|
|
|
s[pos++] = d;
|
|
|
|
if (strchr(enders, d) || (pos >= len)) {
|
|
|
|
s[pos]='\0';
|
1999-12-04 21:35:07 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
to = timeout;
|
2006-04-14 20:47:06 +00:00
|
|
|
}
|
1999-12-04 21:35:07 +00:00
|
|
|
/* Never reached */
|
|
|
|
return 0;
|
|
|
|
}
|
2001-03-12 03:04:51 +00:00
|
|
|
|
2001-09-19 14:40:16 +00:00
|
|
|
int ast_channel_supports_html(struct ast_channel *chan)
|
|
|
|
{
|
2006-04-14 20:47:06 +00:00
|
|
|
return (chan->tech->send_html) ? 1 : 0;
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
|
|
|
|
2005-03-28 20:48:24 +00:00
|
|
|
int ast_channel_sendhtml(struct ast_channel *chan, int subclass, const char *data, int datalen)
|
2001-09-19 14:40:16 +00:00
|
|
|
{
|
2005-03-04 06:47:24 +00:00
|
|
|
if (chan->tech->send_html)
|
|
|
|
return chan->tech->send_html(chan, subclass, data, datalen);
|
2001-09-19 14:40:16 +00:00
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
2005-03-28 20:48:24 +00:00
|
|
|
int ast_channel_sendurl(struct ast_channel *chan, const char *url)
|
2001-09-19 14:40:16 +00:00
|
|
|
{
|
2006-04-14 20:47:06 +00:00
|
|
|
return ast_channel_sendhtml(chan, AST_HTML_URL, url, strlen(url) + 1);
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
|
|
|
|
2007-01-07 14:32:20 +00:00
|
|
|
/*! \brief Set up translation from one channel to another */
|
|
|
|
static int ast_channel_make_compatible_helper(struct ast_channel *from, struct ast_channel *to)
|
2001-03-12 03:04:51 +00:00
|
|
|
{
|
2005-04-18 16:26:41 +00:00
|
|
|
int src;
|
|
|
|
int dst;
|
2005-04-04 03:28:38 +00:00
|
|
|
|
2007-06-12 19:19:09 +00:00
|
|
|
if (from->readformat == to->writeformat && from->writeformat == to->readformat) {
|
|
|
|
/* Already compatible! Moving on ... */
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2007-01-07 14:32:20 +00:00
|
|
|
/* Set up translation from the 'from' channel to the 'to' channel */
|
|
|
|
src = from->nativeformats;
|
|
|
|
dst = to->nativeformats;
|
2008-02-18 17:12:13 +00:00
|
|
|
|
|
|
|
/* If there's no audio in this call, don't bother with trying to find a translation path */
|
|
|
|
if ((src & AST_FORMAT_AUDIO_MASK) == 0 || (dst & AST_FORMAT_AUDIO_MASK) == 0)
|
|
|
|
return 0;
|
|
|
|
|
2005-04-18 16:26:41 +00:00
|
|
|
if (ast_translator_best_choice(&dst, &src) < 0) {
|
2007-01-07 14:32:20 +00:00
|
|
|
ast_log(LOG_WARNING, "No path to translate from %s(%d) to %s(%d)\n", from->name, src, to->name, dst);
|
2001-03-12 03:04:51 +00:00
|
|
|
return -1;
|
|
|
|
}
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2005-04-18 16:26:41 +00:00
|
|
|
/* if the best path is not 'pass through', then
|
|
|
|
transcoding is needed; if desired, force transcode path
|
2006-04-17 16:42:03 +00:00
|
|
|
to use SLINEAR between channels, but only if there is
|
|
|
|
no direct conversion available */
|
2006-04-17 16:43:55 +00:00
|
|
|
if ((src != dst) && ast_opt_transcode_via_slin &&
|
|
|
|
(ast_translate_path_steps(dst, src) != 1))
|
2005-04-18 16:26:41 +00:00
|
|
|
dst = AST_FORMAT_SLINEAR;
|
2007-01-07 14:32:20 +00:00
|
|
|
if (ast_set_read_format(from, dst) < 0) {
|
|
|
|
ast_log(LOG_WARNING, "Unable to set read format on channel %s to %d\n", from->name, dst);
|
2001-03-12 03:04:51 +00:00
|
|
|
return -1;
|
|
|
|
}
|
2007-01-07 14:32:20 +00:00
|
|
|
if (ast_set_write_format(to, dst) < 0) {
|
|
|
|
ast_log(LOG_WARNING, "Unable to set write format on channel %s to %d\n", to->name, dst);
|
2001-03-12 03:04:51 +00:00
|
|
|
return -1;
|
|
|
|
}
|
2007-01-07 14:32:20 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
int ast_channel_make_compatible(struct ast_channel *chan, struct ast_channel *peer)
|
|
|
|
{
|
|
|
|
/* Some callers do not check return code, and we must try to set all call legs correctly */
|
|
|
|
int rc = 0;
|
|
|
|
|
|
|
|
/* Set up translation from the chan to the peer */
|
|
|
|
rc = ast_channel_make_compatible_helper(chan, peer);
|
|
|
|
|
|
|
|
if (rc < 0)
|
|
|
|
return rc;
|
2005-04-18 16:26:41 +00:00
|
|
|
|
|
|
|
/* Set up translation from the peer to the chan */
|
2007-01-07 14:32:20 +00:00
|
|
|
rc = ast_channel_make_compatible_helper(peer, chan);
|
2006-04-17 16:42:03 +00:00
|
|
|
|
2007-01-07 14:32:20 +00:00
|
|
|
return rc;
|
2001-03-12 03:04:51 +00:00
|
|
|
}
|
|
|
|
|
2008-08-10 19:35:50 +00:00
|
|
|
int ast_channel_masquerade(struct ast_channel *original, struct ast_channel *clonechan)
|
2001-05-07 00:43:32 +00:00
|
|
|
{
|
2004-04-20 03:16:01 +00:00
|
|
|
int res = -1;
|
2007-10-01 21:31:16 +00:00
|
|
|
struct ast_channel *final_orig, *final_clone, *base;
|
2007-09-28 05:37:16 +00:00
|
|
|
|
|
|
|
retrymasq:
|
|
|
|
final_orig = original;
|
2008-08-10 19:35:50 +00:00
|
|
|
final_clone = clonechan;
|
2006-07-10 21:07:48 +00:00
|
|
|
|
|
|
|
ast_channel_lock(original);
|
2008-08-10 19:35:50 +00:00
|
|
|
while (ast_channel_trylock(clonechan)) {
|
2006-07-10 21:07:48 +00:00
|
|
|
ast_channel_unlock(original);
|
|
|
|
usleep(1);
|
|
|
|
ast_channel_lock(original);
|
|
|
|
}
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2006-06-01 20:53:17 +00:00
|
|
|
/* each of these channels may be sitting behind a channel proxy (i.e. chan_agent)
|
|
|
|
and if so, we don't really want to masquerade it, but its proxy */
|
2007-02-22 18:53:22 +00:00
|
|
|
if (original->_bridge && (original->_bridge != ast_bridged_channel(original)) && (original->_bridge->_bridge != original))
|
2006-07-10 21:07:48 +00:00
|
|
|
final_orig = original->_bridge;
|
2006-06-01 20:53:17 +00:00
|
|
|
|
2008-08-10 19:35:50 +00:00
|
|
|
if (clonechan->_bridge && (clonechan->_bridge != ast_bridged_channel(clonechan)) && (clonechan->_bridge->_bridge != clonechan))
|
|
|
|
final_clone = clonechan->_bridge;
|
2007-10-01 21:31:16 +00:00
|
|
|
|
|
|
|
if (final_clone->tech->get_base_channel && (base = final_clone->tech->get_base_channel(final_clone))) {
|
|
|
|
final_clone = base;
|
|
|
|
}
|
2006-07-10 21:07:48 +00:00
|
|
|
|
2008-08-10 19:35:50 +00:00
|
|
|
if ((final_orig != original) || (final_clone != clonechan)) {
|
2007-09-28 05:37:16 +00:00
|
|
|
/* Lots and lots of deadlock avoidance. The main one we're competing with
|
|
|
|
* is ast_write(), which locks channels recursively, when working with a
|
|
|
|
* proxy channel. */
|
|
|
|
if (ast_channel_trylock(final_orig)) {
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_channel_unlock(clonechan);
|
2007-09-28 05:37:16 +00:00
|
|
|
ast_channel_unlock(original);
|
|
|
|
goto retrymasq;
|
|
|
|
}
|
|
|
|
if (ast_channel_trylock(final_clone)) {
|
2006-07-10 21:07:48 +00:00
|
|
|
ast_channel_unlock(final_orig);
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_channel_unlock(clonechan);
|
2007-09-28 05:37:16 +00:00
|
|
|
ast_channel_unlock(original);
|
|
|
|
goto retrymasq;
|
2006-07-10 21:07:48 +00:00
|
|
|
}
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_channel_unlock(clonechan);
|
2006-07-10 21:07:48 +00:00
|
|
|
ast_channel_unlock(original);
|
|
|
|
original = final_orig;
|
2008-08-10 19:35:50 +00:00
|
|
|
clonechan = final_clone;
|
2006-07-10 21:07:48 +00:00
|
|
|
}
|
2006-06-01 20:53:17 +00:00
|
|
|
|
2008-08-10 19:35:50 +00:00
|
|
|
if (original == clonechan) {
|
2005-01-09 09:42:54 +00:00
|
|
|
ast_log(LOG_WARNING, "Can't masquerade channel '%s' into itself!\n", original->name);
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_channel_unlock(clonechan);
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_unlock(original);
|
2006-07-10 21:07:48 +00:00
|
|
|
return -1;
|
2004-04-20 03:16:01 +00:00
|
|
|
}
|
2006-07-10 21:07:48 +00:00
|
|
|
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Planning to masquerade channel %s into the structure of %s\n",
|
2008-08-10 19:35:50 +00:00
|
|
|
clonechan->name, original->name);
|
2001-05-07 00:43:32 +00:00
|
|
|
if (original->masq) {
|
2006-04-06 15:40:12 +00:00
|
|
|
ast_log(LOG_WARNING, "%s is already going to masquerade as %s\n",
|
2001-05-07 00:43:32 +00:00
|
|
|
original->masq->name, original->name);
|
2008-08-10 19:35:50 +00:00
|
|
|
} else if (clonechan->masqr) {
|
2006-04-06 15:40:12 +00:00
|
|
|
ast_log(LOG_WARNING, "%s is already going to masquerade as %s\n",
|
2008-08-10 19:35:50 +00:00
|
|
|
clonechan->name, clonechan->masqr->name);
|
2004-04-20 03:16:01 +00:00
|
|
|
} else {
|
2008-08-10 19:35:50 +00:00
|
|
|
original->masq = clonechan;
|
|
|
|
clonechan->masqr = original;
|
2006-04-14 20:47:06 +00:00
|
|
|
ast_queue_frame(original, &ast_null_frame);
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_queue_frame(clonechan, &ast_null_frame);
|
|
|
|
ast_debug(1, "Done planning to masquerade channel %s into the structure of %s\n", clonechan->name, original->name);
|
2004-04-20 03:16:01 +00:00
|
|
|
res = 0;
|
|
|
|
}
|
2006-07-10 21:07:48 +00:00
|
|
|
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_channel_unlock(clonechan);
|
2006-04-26 21:44:25 +00:00
|
|
|
ast_channel_unlock(original);
|
2006-07-10 21:07:48 +00:00
|
|
|
|
2004-04-20 03:16:01 +00:00
|
|
|
return res;
|
2001-05-07 00:43:32 +00:00
|
|
|
}
|
|
|
|
|
2003-02-05 21:18:09 +00:00
|
|
|
void ast_change_name(struct ast_channel *chan, char *newname)
|
|
|
|
{
|
2007-12-06 15:29:05 +00:00
|
|
|
manager_event(EVENT_FLAG_CALL, "Rename", "Channel: %s\r\nNewname: %s\r\nUniqueid: %s\r\n", chan->name, newname, chan->uniqueid);
|
2006-02-01 23:05:28 +00:00
|
|
|
ast_string_field_set(chan, name, newname);
|
2003-02-05 21:18:09 +00:00
|
|
|
}
|
|
|
|
|
2005-01-08 17:23:29 +00:00
|
|
|
void ast_channel_inherit_variables(const struct ast_channel *parent, struct ast_channel *child)
|
|
|
|
{
|
|
|
|
struct ast_var_t *current, *newvar;
|
2005-12-03 19:25:33 +00:00
|
|
|
const char *varname;
|
2005-01-08 17:23:29 +00:00
|
|
|
|
|
|
|
AST_LIST_TRAVERSE(&parent->varshead, current, entries) {
|
|
|
|
int vartype = 0;
|
|
|
|
|
|
|
|
varname = ast_var_full_name(current);
|
|
|
|
if (!varname)
|
|
|
|
continue;
|
|
|
|
|
|
|
|
if (varname[0] == '_') {
|
|
|
|
vartype = 1;
|
|
|
|
if (varname[1] == '_')
|
|
|
|
vartype = 2;
|
|
|
|
}
|
|
|
|
|
|
|
|
switch (vartype) {
|
|
|
|
case 1:
|
|
|
|
newvar = ast_var_assign(&varname[1], ast_var_value(current));
|
|
|
|
if (newvar) {
|
2005-09-01 21:50:49 +00:00
|
|
|
AST_LIST_INSERT_TAIL(&child->varshead, newvar, entries);
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Copying soft-transferable variable %s.\n", ast_var_name(newvar));
|
2005-01-08 17:23:29 +00:00
|
|
|
}
|
|
|
|
break;
|
|
|
|
case 2:
|
2007-12-03 20:30:02 +00:00
|
|
|
newvar = ast_var_assign(varname, ast_var_value(current));
|
2005-01-08 17:23:29 +00:00
|
|
|
if (newvar) {
|
2005-09-01 21:50:49 +00:00
|
|
|
AST_LIST_INSERT_TAIL(&child->varshead, newvar, entries);
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Copying hard-transferable variable %s.\n", ast_var_name(newvar));
|
2005-01-08 17:23:29 +00:00
|
|
|
}
|
|
|
|
break;
|
|
|
|
default:
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Not copying variable %s.\n", ast_var_name(current));
|
2005-01-08 17:23:29 +00:00
|
|
|
break;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*!
|
|
|
|
\brief Clone channel variables from 'clone' channel into 'original' channel
|
2006-04-06 15:40:12 +00:00
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
All variables except those related to app_groupcount are cloned.
|
|
|
|
Variables are actually _removed_ from 'clone' channel, presumably
|
|
|
|
because it will subsequently be destroyed.
|
2006-04-06 15:40:12 +00:00
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
\note Assumes locks will be in place on both channels when called.
|
|
|
|
*/
|
2008-08-10 19:35:50 +00:00
|
|
|
static void clone_variables(struct ast_channel *original, struct ast_channel *clonechan)
|
2005-01-02 02:07:24 +00:00
|
|
|
{
|
2007-06-19 18:31:29 +00:00
|
|
|
struct ast_var_t *current, *newvar;
|
2005-01-02 02:07:24 +00:00
|
|
|
/* Append variables from clone channel into original channel */
|
|
|
|
/* XXX Is this always correct? We have to in order to keep MACROS working XXX */
|
2008-08-10 19:35:50 +00:00
|
|
|
if (AST_LIST_FIRST(&clonechan->varshead))
|
|
|
|
AST_LIST_APPEND_LIST(&original->varshead, &clonechan->varshead, entries);
|
2007-06-19 18:31:29 +00:00
|
|
|
|
|
|
|
/* then, dup the varshead list into the clone */
|
|
|
|
|
|
|
|
AST_LIST_TRAVERSE(&original->varshead, current, entries) {
|
|
|
|
newvar = ast_var_assign(current->name, current->value);
|
|
|
|
if (newvar)
|
2008-08-10 19:35:50 +00:00
|
|
|
AST_LIST_INSERT_TAIL(&clonechan->varshead, newvar, entries);
|
2007-06-19 18:31:29 +00:00
|
|
|
}
|
2005-01-02 02:07:24 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*!
|
|
|
|
\brief Masquerade a channel
|
|
|
|
|
2006-04-06 15:40:12 +00:00
|
|
|
\note Assumes channel will be locked when called
|
2005-12-31 16:35:48 +00:00
|
|
|
*/
|
2004-04-06 22:17:32 +00:00
|
|
|
int ast_do_masquerade(struct ast_channel *original)
|
2001-05-07 00:43:32 +00:00
|
|
|
{
|
2003-08-14 19:18:18 +00:00
|
|
|
int x,i;
|
2001-05-07 00:43:32 +00:00
|
|
|
int res=0;
|
2004-07-09 10:57:43 +00:00
|
|
|
int origstate;
|
2008-08-10 19:35:50 +00:00
|
|
|
struct ast_frame *current;
|
2005-03-04 06:47:24 +00:00
|
|
|
const struct ast_channel_tech *t;
|
|
|
|
void *t_pvt;
|
2004-10-02 00:58:31 +00:00
|
|
|
struct ast_callerid tmpcid;
|
2008-08-10 19:35:50 +00:00
|
|
|
struct ast_channel *clonechan = original->masq;
|
2007-03-30 14:37:21 +00:00
|
|
|
struct ast_cdr *cdr;
|
2001-09-19 14:40:16 +00:00
|
|
|
int rformat = original->readformat;
|
|
|
|
int wformat = original->writeformat;
|
2008-01-03 21:27:57 +00:00
|
|
|
char newn[AST_CHANNEL_NAME];
|
|
|
|
char orig[AST_CHANNEL_NAME];
|
|
|
|
char masqn[AST_CHANNEL_NAME];
|
|
|
|
char zombn[AST_CHANNEL_NAME];
|
2005-01-10 04:23:39 +00:00
|
|
|
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(4, "Actually Masquerading %s(%d) into the structure of %s(%d)\n",
|
2008-08-10 19:35:50 +00:00
|
|
|
clonechan->name, clonechan->_state, original->name, original->_state);
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2006-10-03 17:10:16 +00:00
|
|
|
manager_event(EVENT_FLAG_CALL, "Masquerade", "Clone: %s\r\nCloneState: %s\r\nOriginal: %s\r\nOriginalState: %s\r\n",
|
2008-08-10 19:35:50 +00:00
|
|
|
clonechan->name, ast_state2str(clonechan->_state), original->name, ast_state2str(original->_state));
|
2006-10-03 17:10:16 +00:00
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
|
/* XXX This is a seriously wacked out operation. We're essentially putting the guts of
|
2004-04-27 21:21:57 +00:00
|
|
|
the clone channel into the original channel. Start by killing off the original
|
2006-04-06 15:40:12 +00:00
|
|
|
channel's backend. I'm not sure we're going to keep this function, because
|
2001-05-07 00:43:32 +00:00
|
|
|
while the features are nice, the cost is very high in terms of pure nastiness. XXX */
|
|
|
|
|
2004-04-06 22:17:32 +00:00
|
|
|
/* We need the clone's lock, too */
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_channel_lock(clonechan);
|
2003-08-11 21:31:53 +00:00
|
|
|
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_debug(2, "Got clone lock for masquerade on '%s' at %p\n", clonechan->name, &clonechan->lock_dont_use);
|
2003-08-11 21:31:53 +00:00
|
|
|
|
2001-09-19 14:40:16 +00:00
|
|
|
/* Having remembered the original read/write formats, we turn off any translation on either
|
|
|
|
one */
|
2008-08-10 19:35:50 +00:00
|
|
|
free_translation(clonechan);
|
2001-09-19 14:40:16 +00:00
|
|
|
free_translation(original);
|
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
|
|
|
|
|
/* Unlink the masquerade */
|
|
|
|
original->masq = NULL;
|
2008-08-10 19:35:50 +00:00
|
|
|
clonechan->masqr = NULL;
|
2002-09-10 20:48:20 +00:00
|
|
|
|
|
|
|
/* Save the original name */
|
2005-06-05 16:32:16 +00:00
|
|
|
ast_copy_string(orig, original->name, sizeof(orig));
|
2002-09-10 20:48:20 +00:00
|
|
|
/* Save the new name */
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_copy_string(newn, clonechan->name, sizeof(newn));
|
2002-09-10 20:48:20 +00:00
|
|
|
/* Create the masq name */
|
|
|
|
snprintf(masqn, sizeof(masqn), "%s<MASQ>", newn);
|
2001-05-07 00:43:32 +00:00
|
|
|
|
|
|
|
/* Copy the name from the clone channel */
|
2006-02-01 23:05:28 +00:00
|
|
|
ast_string_field_set(original, name, newn);
|
2001-05-07 00:43:32 +00:00
|
|
|
|
|
|
|
/* Mangle the name of the clone channel */
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_string_field_set(clonechan, name, masqn);
|
2002-09-10 20:48:20 +00:00
|
|
|
|
|
|
|
/* Notify any managers of the change, first the masq then the other */
|
2008-08-10 19:35:50 +00:00
|
|
|
manager_event(EVENT_FLAG_CALL, "Rename", "Channel: %s\r\nNewname: %s\r\nUniqueid: %s\r\n", newn, masqn, clonechan->uniqueid);
|
2007-12-06 15:29:05 +00:00
|
|
|
manager_event(EVENT_FLAG_CALL, "Rename", "Channel: %s\r\nNewname: %s\r\nUniqueid: %s\r\n", orig, newn, original->uniqueid);
|
2001-05-07 00:43:32 +00:00
|
|
|
|
2006-01-16 17:37:44 +00:00
|
|
|
/* Swap the technologies */
|
2005-03-04 06:47:24 +00:00
|
|
|
t = original->tech;
|
2008-08-10 19:35:50 +00:00
|
|
|
original->tech = clonechan->tech;
|
|
|
|
clonechan->tech = t;
|
2005-03-04 06:47:24 +00:00
|
|
|
|
2007-03-30 14:37:21 +00:00
|
|
|
/* Swap the cdrs */
|
|
|
|
cdr = original->cdr;
|
2008-08-10 19:35:50 +00:00
|
|
|
original->cdr = clonechan->cdr;
|
|
|
|
clonechan->cdr = cdr;
|
2007-03-30 14:37:21 +00:00
|
|
|
|
2005-03-04 06:47:24 +00:00
|
|
|
t_pvt = original->tech_pvt;
|
2008-08-10 19:35:50 +00:00
|
|
|
original->tech_pvt = clonechan->tech_pvt;
|
|
|
|
clonechan->tech_pvt = t_pvt;
|
2005-03-04 06:47:24 +00:00
|
|
|
|
|
|
|
/* Swap the alertpipes */
|
|
|
|
for (i = 0; i < 2; i++) {
|
|
|
|
x = original->alertpipe[i];
|
2008-08-10 19:35:50 +00:00
|
|
|
original->alertpipe[i] = clonechan->alertpipe[i];
|
|
|
|
clonechan->alertpipe[i] = x;
|
2005-03-04 06:47:24 +00:00
|
|
|
}
|
|
|
|
|
2007-11-05 20:14:48 +00:00
|
|
|
/*
|
|
|
|
* Swap the readq's. The end result should be this:
|
|
|
|
*
|
|
|
|
* 1) All frames should be on the new (original) channel.
|
|
|
|
* 2) Any frames that were already on the new channel before this
|
|
|
|
* masquerade need to be at the end of the readq, after all of the
|
|
|
|
* frames on the old (clone) channel.
|
|
|
|
* 3) The alertpipe needs to get poked for every frame that was already
|
|
|
|
* on the new channel, since we are now using the alert pipe from the
|
|
|
|
* old (clone) channel.
|
|
|
|
*/
|
|
|
|
{
|
|
|
|
AST_LIST_HEAD_NOLOCK(, ast_frame) tmp_readq;
|
|
|
|
AST_LIST_HEAD_SET_NOLOCK(&tmp_readq, NULL);
|
|
|
|
|
|
|
|
AST_LIST_APPEND_LIST(&tmp_readq, &original->readq, frame_list);
|
2008-08-10 19:35:50 +00:00
|
|
|
AST_LIST_APPEND_LIST(&original->readq, &clonechan->readq, frame_list);
|
2007-11-05 20:14:48 +00:00
|
|
|
|
2008-08-10 19:35:50 +00:00
|
|
|
while ((current = AST_LIST_REMOVE_HEAD(&tmp_readq, frame_list))) {
|
|
|
|
AST_LIST_INSERT_TAIL(&original->readq, current, frame_list);
|
2007-11-05 20:14:48 +00:00
|
|
|
if (original->alertpipe[1] > -1) {
|
|
|
|
int poke = 0;
|
2008-11-02 18:52:13 +00:00
|
|
|
|
|
|
|
if (write(original->alertpipe[1], &poke, sizeof(poke)) < 0) {
|
|
|
|
ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
|
|
|
|
}
|
2007-11-05 20:14:48 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2005-03-04 06:47:24 +00:00
|
|
|
/* Swap the raw formats */
|
|
|
|
x = original->rawreadformat;
|
2008-08-10 19:35:50 +00:00
|
|
|
original->rawreadformat = clonechan->rawreadformat;
|
|
|
|
clonechan->rawreadformat = x;
|
2005-03-04 06:47:24 +00:00
|
|
|
x = original->rawwriteformat;
|
2008-08-10 19:35:50 +00:00
|
|
|
original->rawwriteformat = clonechan->rawwriteformat;
|
|
|
|
clonechan->rawwriteformat = x;
|
2003-08-14 19:18:18 +00:00
|
|
|
|
2008-08-10 19:35:50 +00:00
|
|
|
clonechan->_softhangup = AST_SOFTHANGUP_DEV;
|
2001-09-19 14:40:16 +00:00
|
|
|
|
2004-07-09 10:57:43 +00:00
|
|
|
/* And of course, so does our current state. Note we need not
|
|
|
|
call ast_setstate since the event manager doesn't really consider
|
|
|
|
these separate. We do this early so that the clone has the proper
|
|
|
|
state of the original channel. */
|
|
|
|
origstate = original->_state;
|
2008-08-10 19:35:50 +00:00
|
|
|
original->_state = clonechan->_state;
|
|
|
|
clonechan->_state = origstate;
|
2004-07-09 10:57:43 +00:00
|
|
|
|
2008-08-10 19:35:50 +00:00
|
|
|
if (clonechan->tech->fixup){
|
|
|
|
res = clonechan->tech->fixup(original, clonechan);
|
2006-04-06 15:40:12 +00:00
|
|
|
if (res)
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_log(LOG_WARNING, "Fixup failed on channel %s, strange things may happen.\n", clonechan->name);
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
2001-05-07 00:43:32 +00:00
|
|
|
|
|
|
|
/* Start by disconnecting the original's physical side */
|
2008-08-10 19:35:50 +00:00
|
|
|
if (clonechan->tech->hangup)
|
|
|
|
res = clonechan->tech->hangup(clonechan);
|
2001-05-07 00:43:32 +00:00
|
|
|
if (res) {
|
|
|
|
ast_log(LOG_WARNING, "Hangup failed! Strange things may happen!\n");
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_channel_unlock(clonechan);
|
2001-05-07 00:43:32 +00:00
|
|
|
return -1;
|
|
|
|
}
|
2008-01-03 21:27:57 +00:00
|
|
|
|
2002-09-10 20:48:20 +00:00
|
|
|
snprintf(zombn, sizeof(zombn), "%s<ZOMBIE>", orig);
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Mangle the name of the clone channel */
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_string_field_set(clonechan, name, zombn);
|
|
|
|
manager_event(EVENT_FLAG_CALL, "Rename", "Channel: %s\r\nNewname: %s\r\nUniqueid: %s\r\n", masqn, zombn, clonechan->uniqueid);
|
2001-05-07 00:43:32 +00:00
|
|
|
|
|
|
|
/* Update the type. */
|
2005-07-31 21:27:21 +00:00
|
|
|
t_pvt = original->monitor;
|
2008-08-10 19:35:50 +00:00
|
|
|
original->monitor = clonechan->monitor;
|
|
|
|
clonechan->monitor = t_pvt;
|
2008-01-03 21:27:57 +00:00
|
|
|
|
2005-01-21 22:30:16 +00:00
|
|
|
/* Keep the same language. */
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_string_field_set(original, language, clonechan->language);
|
2006-01-30 03:13:33 +00:00
|
|
|
/* Copy the FD's other than the generator fd */
|
2005-07-25 22:13:32 +00:00
|
|
|
for (x = 0; x < AST_MAX_FDS; x++) {
|
2006-01-30 03:13:33 +00:00
|
|
|
if (x != AST_GENERATOR_FD)
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_channel_set_fd(original, x, clonechan->fds[x]);
|
2003-06-29 23:23:54 +00:00
|
|
|
}
|
2006-07-28 18:59:59 +00:00
|
|
|
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_app_group_update(clonechan, original);
|
2007-04-25 19:27:42 +00:00
|
|
|
|
2006-04-10 23:29:50 +00:00
|
|
|
/* Move data stores over */
|
2008-08-10 19:35:50 +00:00
|
|
|
if (AST_LIST_FIRST(&clonechan->datastores)) {
|
2008-03-13 21:40:43 +00:00
|
|
|
struct ast_datastore *ds;
|
2008-08-10 19:35:50 +00:00
|
|
|
AST_LIST_APPEND_LIST(&original->datastores, &clonechan->datastores, entry);
|
2008-03-13 21:40:43 +00:00
|
|
|
AST_LIST_TRAVERSE(&original->datastores, ds, entry) {
|
|
|
|
if (ds->info->chan_fixup)
|
2008-08-10 19:35:50 +00:00
|
|
|
ds->info->chan_fixup(ds->data, clonechan, original);
|
2008-03-13 21:40:43 +00:00
|
|
|
}
|
|
|
|
}
|
2006-04-10 23:29:50 +00:00
|
|
|
|
2008-08-10 19:35:50 +00:00
|
|
|
clone_variables(original, clonechan);
|
2001-12-20 15:21:47 +00:00
|
|
|
/* Presense of ADSI capable CPE follows clone */
|
2008-08-10 19:35:50 +00:00
|
|
|
original->adsicpe = clonechan->adsicpe;
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Bridge remains the same */
|
|
|
|
/* CDR fields remain the same */
|
|
|
|
/* XXX What about blocking, softhangup, blocker, and lock and blockproc? XXX */
|
|
|
|
/* Application and data remain the same */
|
2004-04-27 21:21:57 +00:00
|
|
|
/* Clone exception becomes real one, as with fdno */
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_copy_flags(original, clonechan, AST_FLAG_EXCEPTION);
|
|
|
|
original->fdno = clonechan->fdno;
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Schedule context remains the same */
|
|
|
|
/* Stream stuff stays the same */
|
2004-04-27 21:21:57 +00:00
|
|
|
/* Keep the original state. The fixup code will need to work with it most likely */
|
2001-05-07 00:43:32 +00:00
|
|
|
|
2004-10-02 00:58:31 +00:00
|
|
|
/* Just swap the whole structures, nevermind the allocations, they'll work themselves
|
|
|
|
out. */
|
|
|
|
tmpcid = original->cid;
|
2008-08-10 19:35:50 +00:00
|
|
|
original->cid = clonechan->cid;
|
|
|
|
clonechan->cid = tmpcid;
|
2008-01-03 21:27:57 +00:00
|
|
|
|
2003-06-29 23:23:54 +00:00
|
|
|
/* Restore original timing file descriptor */
|
2007-08-08 21:44:58 +00:00
|
|
|
ast_channel_set_fd(original, AST_TIMING_FD, original->timingfd);
|
2008-01-03 21:27:57 +00:00
|
|
|
|
2001-09-19 14:40:16 +00:00
|
|
|
/* Our native formats are different now */
|
2008-08-10 19:35:50 +00:00
|
|
|
original->nativeformats = clonechan->nativeformats;
|
2008-01-03 21:27:57 +00:00
|
|
|
|
2004-04-27 21:21:57 +00:00
|
|
|
/* Context, extension, priority, app data, jump table, remain the same */
|
2001-05-07 00:43:32 +00:00
|
|
|
/* pvt switches. pbx stays the same, as does next */
|
2008-01-03 21:27:57 +00:00
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Set the write format */
|
2004-04-06 22:17:32 +00:00
|
|
|
ast_set_write_format(original, wformat);
|
2001-05-07 00:43:32 +00:00
|
|
|
|
|
|
|
/* Set the read format */
|
2004-04-06 22:17:32 +00:00
|
|
|
ast_set_read_format(original, rformat);
|
2001-09-19 14:40:16 +00:00
|
|
|
|
2005-01-10 04:23:39 +00:00
|
|
|
/* Copy the music class */
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_string_field_set(original, musicclass, clonechan->musicclass);
|
2005-01-10 04:23:39 +00:00
|
|
|
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Putting channel %s in %d/%d formats\n", original->name, wformat, rformat);
|
2001-03-12 03:04:51 +00:00
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Okay. Last thing is to let the channel driver know about all this mess, so he
|
|
|
|
can fix up everything as best as possible */
|
2005-03-04 06:47:24 +00:00
|
|
|
if (original->tech->fixup) {
|
2008-08-10 19:35:50 +00:00
|
|
|
res = original->tech->fixup(clonechan, original);
|
2001-05-07 00:43:32 +00:00
|
|
|
if (res) {
|
2005-07-25 22:13:32 +00:00
|
|
|
ast_log(LOG_WARNING, "Channel for type '%s' could not fixup channel %s\n",
|
2006-02-01 23:05:28 +00:00
|
|
|
original->tech->type, original->name);
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_channel_unlock(clonechan);
|
2001-05-07 00:43:32 +00:00
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
} else
|
2005-07-25 22:13:32 +00:00
|
|
|
ast_log(LOG_WARNING, "Channel type '%s' does not have a fixup routine (for %s)! Bad things may happen.\n",
|
2006-02-01 23:05:28 +00:00
|
|
|
original->tech->type, original->name);
|
2007-12-03 18:44:16 +00:00
|
|
|
|
|
|
|
/* If an indication is currently playing maintain it on the channel that is taking the place of original */
|
|
|
|
if (original->visible_indication)
|
|
|
|
ast_indicate(original, original->visible_indication);
|
2008-01-03 21:27:57 +00:00
|
|
|
|
2004-02-19 20:07:01 +00:00
|
|
|
/* Now, at this point, the "clone" channel is totally F'd up. We mark it as
|
|
|
|
a zombie so nothing tries to touch it. If it's already been marked as a
|
|
|
|
zombie, then free it now (since it already is considered invalid). */
|
2008-08-10 19:35:50 +00:00
|
|
|
if (ast_test_flag(clonechan, AST_FLAG_ZOMBIE)) {
|
|
|
|
ast_debug(1, "Destroying channel clone '%s'\n", clonechan->name);
|
|
|
|
ast_channel_unlock(clonechan);
|
2006-04-06 15:40:12 +00:00
|
|
|
manager_event(EVENT_FLAG_CALL, "Hangup",
|
2005-07-25 22:13:32 +00:00
|
|
|
"Channel: %s\r\n"
|
|
|
|
"Uniqueid: %s\r\n"
|
|
|
|
"Cause: %d\r\n"
|
|
|
|
"Cause-txt: %s\r\n",
|
2008-08-10 19:35:50 +00:00
|
|
|
clonechan->name,
|
|
|
|
clonechan->uniqueid,
|
|
|
|
clonechan->hangupcause,
|
|
|
|
ast_cause2str(clonechan->hangupcause)
|
2005-07-25 22:13:32 +00:00
|
|
|
);
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_channel_free(clonechan);
|
2004-02-19 20:07:01 +00:00
|
|
|
} else {
|
2008-08-10 19:35:50 +00:00
|
|
|
ast_debug(1, "Released clone lock on '%s'\n", clonechan->name);
|
|
|
|
ast_set_flag(clonechan, AST_FLAG_ZOMBIE);
|
|
|
|
ast_queue_frame(clonechan, &ast_null_frame);
|
|
|
|
ast_channel_unlock(clonechan);
|
2004-02-19 20:07:01 +00:00
|
|
|
}
|
2008-01-03 21:27:57 +00:00
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
|
/* Signal any blocker */
|
2004-12-07 20:38:43 +00:00
|
|
|
if (ast_test_flag(original, AST_FLAG_BLOCKING))
|
2001-05-07 00:43:32 +00:00
|
|
|
pthread_kill(original->blocker, SIGURG);
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Done Masquerading %s (%d)\n", original->name, original->_state);
|
2002-09-10 20:48:20 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2007-11-25 19:33:33 +00:00
|
|
|
void ast_set_callerid(struct ast_channel *chan, const char *cid_num, const char *cid_name, const char *cid_ani)
|
2002-09-10 20:48:20 +00:00
|
|
|
{
|
2007-11-29 00:28:10 +00:00
|
|
|
ast_channel_lock(chan);
|
|
|
|
|
2007-11-25 19:33:33 +00:00
|
|
|
if (cid_num) {
|
2004-10-02 00:58:31 +00:00
|
|
|
if (chan->cid.cid_num)
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(chan->cid.cid_num);
|
2007-11-25 19:33:33 +00:00
|
|
|
chan->cid.cid_num = ast_strdup(cid_num);
|
2004-10-02 00:58:31 +00:00
|
|
|
}
|
2007-11-25 19:33:33 +00:00
|
|
|
if (cid_name) {
|
2004-10-02 00:58:31 +00:00
|
|
|
if (chan->cid.cid_name)
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(chan->cid.cid_name);
|
2007-11-25 19:33:33 +00:00
|
|
|
chan->cid.cid_name = ast_strdup(cid_name);
|
2004-10-02 00:58:31 +00:00
|
|
|
}
|
2007-11-25 19:33:33 +00:00
|
|
|
if (cid_ani) {
|
2004-10-02 00:58:31 +00:00
|
|
|
if (chan->cid.cid_ani)
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(chan->cid.cid_ani);
|
2007-11-25 19:33:33 +00:00
|
|
|
chan->cid.cid_ani = ast_strdup(cid_ani);
|
2003-02-05 21:18:09 +00:00
|
|
|
}
|
2007-12-06 15:29:05 +00:00
|
|
|
manager_event(EVENT_FLAG_CALL, "NewCallerid",
|
2002-09-10 20:48:20 +00:00
|
|
|
"Channel: %s\r\n"
|
2006-10-02 20:35:16 +00:00
|
|
|
"CallerIDNum: %s\r\n"
|
2004-12-21 21:46:58 +00:00
|
|
|
"CallerIDName: %s\r\n"
|
2005-07-25 22:13:32 +00:00
|
|
|
"Uniqueid: %s\r\n"
|
|
|
|
"CID-CallingPres: %d (%s)\r\n",
|
2006-04-21 10:05:45 +00:00
|
|
|
chan->name,
|
2007-12-06 15:29:05 +00:00
|
|
|
S_OR(chan->cid.cid_num, ""),
|
|
|
|
S_OR(chan->cid.cid_name, ""),
|
2005-07-25 22:13:32 +00:00
|
|
|
chan->uniqueid,
|
|
|
|
chan->cid.cid_pres,
|
|
|
|
ast_describe_caller_presentation(chan->cid.cid_pres)
|
|
|
|
);
|
2007-11-29 00:28:10 +00:00
|
|
|
|
|
|
|
ast_channel_unlock(chan);
|
2002-09-10 20:48:20 +00:00
|
|
|
}
|
|
|
|
|
2006-08-19 00:33:44 +00:00
|
|
|
int ast_setstate(struct ast_channel *chan, enum ast_channel_state state)
|
2002-09-10 20:48:20 +00:00
|
|
|
{
|
2005-09-01 22:28:33 +00:00
|
|
|
int oldstate = chan->_state;
|
2008-07-25 17:24:43 +00:00
|
|
|
char name[AST_CHANNEL_NAME], *dashptr;
|
2005-09-01 22:28:33 +00:00
|
|
|
|
|
|
|
if (oldstate == state)
|
|
|
|
return 0;
|
|
|
|
|
2008-07-25 17:24:43 +00:00
|
|
|
ast_copy_string(name, chan->name, sizeof(name));
|
|
|
|
if ((dashptr = strrchr(name, '-'))) {
|
|
|
|
*dashptr = '\0';
|
|
|
|
}
|
|
|
|
|
2005-09-01 22:28:33 +00:00
|
|
|
chan->_state = state;
|
2008-07-25 21:20:03 +00:00
|
|
|
ast_devstate_changed_literal(ast_state_chan2dev(state), name);
|
2006-11-07 21:47:49 +00:00
|
|
|
/* setstate used to conditionally report Newchannel; this is no more */
|
2006-08-17 16:37:39 +00:00
|
|
|
manager_event(EVENT_FLAG_CALL,
|
2006-11-07 21:47:49 +00:00
|
|
|
"Newstate",
|
2005-09-01 22:28:33 +00:00
|
|
|
"Channel: %s\r\n"
|
2007-12-06 15:29:05 +00:00
|
|
|
"ChannelState: %d\r\n"
|
|
|
|
"ChannelStateDesc: %s\r\n"
|
2006-10-02 20:35:16 +00:00
|
|
|
"CallerIDNum: %s\r\n"
|
2005-09-01 22:28:33 +00:00
|
|
|
"CallerIDName: %s\r\n"
|
|
|
|
"Uniqueid: %s\r\n",
|
2007-12-06 15:29:05 +00:00
|
|
|
chan->name, chan->_state, ast_state2str(chan->_state),
|
|
|
|
S_OR(chan->cid.cid_num, ""),
|
|
|
|
S_OR(chan->cid.cid_name, ""),
|
2005-09-01 22:28:33 +00:00
|
|
|
chan->uniqueid);
|
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
|
return 0;
|
|
|
|
}
|
2001-03-12 03:04:51 +00:00
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Find bridged channel */
|
2004-10-23 12:19:47 +00:00
|
|
|
struct ast_channel *ast_bridged_channel(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
struct ast_channel *bridged;
|
|
|
|
bridged = chan->_bridge;
|
2006-04-06 15:40:12 +00:00
|
|
|
if (bridged && bridged->tech->bridged_channel)
|
2005-03-04 06:47:24 +00:00
|
|
|
bridged = bridged->tech->bridged_channel(chan, bridged);
|
2004-10-23 12:19:47 +00:00
|
|
|
return bridged;
|
|
|
|
}
|
|
|
|
|
2006-04-06 15:40:12 +00:00
|
|
|
static void bridge_playfile(struct ast_channel *chan, struct ast_channel *peer, const char *sound, int remain)
|
2004-05-03 02:02:27 +00:00
|
|
|
{
|
2005-12-24 05:18:40 +00:00
|
|
|
int min = 0, sec = 0, check;
|
2004-05-06 22:29:00 +00:00
|
|
|
|
|
|
|
check = ast_autoservice_start(peer);
|
2006-04-06 15:40:12 +00:00
|
|
|
if (check)
|
2004-05-06 22:29:00 +00:00
|
|
|
return;
|
|
|
|
|
2004-05-03 02:02:27 +00:00
|
|
|
if (remain > 0) {
|
|
|
|
if (remain / 60 > 1) {
|
2004-04-26 23:22:34 +00:00
|
|
|
min = remain / 60;
|
|
|
|
sec = remain % 60;
|
2004-05-03 02:02:27 +00:00
|
|
|
} else {
|
2004-04-26 23:22:34 +00:00
|
|
|
sec = remain;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2005-05-15 04:48:30 +00:00
|
|
|
if (!strcmp(sound,"timeleft")) { /* Queue support */
|
2006-11-17 23:18:51 +00:00
|
|
|
ast_stream_and_wait(chan, "vm-youhave", "");
|
2004-05-03 02:02:27 +00:00
|
|
|
if (min) {
|
2006-04-14 20:47:06 +00:00
|
|
|
ast_say_number(chan, min, AST_DIGIT_ANY, chan->language, NULL);
|
2006-11-17 23:18:51 +00:00
|
|
|
ast_stream_and_wait(chan, "queue-minutes", "");
|
2004-04-26 23:22:34 +00:00
|
|
|
}
|
2004-05-03 02:02:27 +00:00
|
|
|
if (sec) {
|
2006-04-14 20:47:06 +00:00
|
|
|
ast_say_number(chan, sec, AST_DIGIT_ANY, chan->language, NULL);
|
2006-11-17 23:18:51 +00:00
|
|
|
ast_stream_and_wait(chan, "queue-seconds", "");
|
2004-04-26 23:22:34 +00:00
|
|
|
}
|
2004-05-03 02:02:27 +00:00
|
|
|
} else {
|
2006-11-17 23:18:51 +00:00
|
|
|
ast_stream_and_wait(chan, sound, "");
|
2004-04-26 23:22:34 +00:00
|
|
|
}
|
2004-05-06 22:29:00 +00:00
|
|
|
|
2006-04-21 20:39:50 +00:00
|
|
|
ast_autoservice_stop(peer);
|
2004-04-26 23:22:34 +00:00
|
|
|
}
|
|
|
|
|
2005-10-13 18:27:54 +00:00
|
|
|
static enum ast_bridge_result ast_generic_bridge(struct ast_channel *c0, struct ast_channel *c1,
|
2006-01-05 23:52:34 +00:00
|
|
|
struct ast_bridge_config *config, struct ast_frame **fo,
|
|
|
|
struct ast_channel **rc, struct timeval bridge_end)
|
2004-05-03 02:02:27 +00:00
|
|
|
{
|
2005-08-09 01:59:59 +00:00
|
|
|
/* Copy voice back and forth between the two channels. */
|
2001-03-12 03:04:51 +00:00
|
|
|
struct ast_channel *cs[3];
|
|
|
|
struct ast_frame *f;
|
2006-05-11 10:37:39 +00:00
|
|
|
enum ast_bridge_result res = AST_BRIDGE_COMPLETE;
|
2005-03-31 03:00:37 +00:00
|
|
|
int o0nativeformats;
|
|
|
|
int o1nativeformats;
|
2005-08-09 01:59:59 +00:00
|
|
|
int watch_c0_dtmf;
|
|
|
|
int watch_c1_dtmf;
|
2005-09-01 00:10:49 +00:00
|
|
|
void *pvt0, *pvt1;
|
2006-05-31 16:56:50 +00:00
|
|
|
/* Indicates whether a frame was queued into a jitterbuffer */
|
|
|
|
int frame_put_in_jb = 0;
|
|
|
|
int jb_in_use;
|
2006-01-05 23:52:34 +00:00
|
|
|
int to;
|
2005-03-31 03:00:37 +00:00
|
|
|
|
|
|
|
cs[0] = c0;
|
|
|
|
cs[1] = c1;
|
2005-09-01 00:10:49 +00:00
|
|
|
pvt0 = c0->tech_pvt;
|
|
|
|
pvt1 = c1->tech_pvt;
|
2005-03-31 03:00:37 +00:00
|
|
|
o0nativeformats = c0->nativeformats;
|
|
|
|
o1nativeformats = c1->nativeformats;
|
2005-08-09 01:59:59 +00:00
|
|
|
watch_c0_dtmf = config->flags & AST_BRIDGE_DTMF_CHANNEL_0;
|
|
|
|
watch_c1_dtmf = config->flags & AST_BRIDGE_DTMF_CHANNEL_1;
|
2005-03-31 03:00:37 +00:00
|
|
|
|
2006-05-31 16:56:50 +00:00
|
|
|
/* Check the need of a jitterbuffer for each channel */
|
|
|
|
jb_in_use = ast_jb_do_usecheck(c0, c1);
|
2008-08-06 00:30:53 +00:00
|
|
|
if (jb_in_use)
|
|
|
|
ast_jb_empty_and_reset(c0, c1);
|
2006-05-31 16:56:50 +00:00
|
|
|
|
2007-08-08 21:44:58 +00:00
|
|
|
ast_poll_channel_add(c0, c1);
|
|
|
|
|
2005-03-31 03:00:37 +00:00
|
|
|
for (;;) {
|
2006-04-16 15:51:53 +00:00
|
|
|
struct ast_channel *who, *other;
|
|
|
|
|
2005-09-01 00:10:49 +00:00
|
|
|
if ((c0->tech_pvt != pvt0) || (c1->tech_pvt != pvt1) ||
|
2005-08-09 01:59:59 +00:00
|
|
|
(o0nativeformats != c0->nativeformats) ||
|
|
|
|
(o1nativeformats != c1->nativeformats)) {
|
2005-03-31 03:00:37 +00:00
|
|
|
/* Check for Masquerade, codec changes, etc */
|
2005-08-09 01:59:59 +00:00
|
|
|
res = AST_BRIDGE_RETRY;
|
2005-03-31 03:00:37 +00:00
|
|
|
break;
|
|
|
|
}
|
2006-01-25 09:50:28 +00:00
|
|
|
if (bridge_end.tv_sec) {
|
|
|
|
to = ast_tvdiff_ms(bridge_end, ast_tvnow());
|
|
|
|
if (to <= 0) {
|
2006-11-15 21:36:13 +00:00
|
|
|
if (config->timelimit)
|
|
|
|
res = AST_BRIDGE_RETRY;
|
|
|
|
else
|
|
|
|
res = AST_BRIDGE_COMPLETE;
|
2006-01-25 09:50:28 +00:00
|
|
|
break;
|
|
|
|
}
|
|
|
|
} else
|
|
|
|
to = -1;
|
2006-05-31 16:56:50 +00:00
|
|
|
/* Calculate the appropriate max sleep interval - in general, this is the time,
|
|
|
|
left to the closest jb delivery moment */
|
|
|
|
if (jb_in_use)
|
|
|
|
to = ast_jb_get_when_to_wakeup(c0, c1, to);
|
2006-01-05 23:52:34 +00:00
|
|
|
who = ast_waitfor_n(cs, 2, &to);
|
2005-03-31 03:00:37 +00:00
|
|
|
if (!who) {
|
2006-05-31 16:56:50 +00:00
|
|
|
/* No frame received within the specified timeout - check if we have to deliver now */
|
|
|
|
if (jb_in_use)
|
|
|
|
ast_jb_get_and_deliver(c0, c1);
|
2005-05-15 04:48:30 +00:00
|
|
|
if (c0->_softhangup == AST_SOFTHANGUP_UNBRIDGE || c1->_softhangup == AST_SOFTHANGUP_UNBRIDGE) {
|
|
|
|
if (c0->_softhangup == AST_SOFTHANGUP_UNBRIDGE)
|
2005-10-14 17:02:20 +00:00
|
|
|
c0->_softhangup = 0;
|
|
|
|
if (c1->_softhangup == AST_SOFTHANGUP_UNBRIDGE)
|
|
|
|
c1->_softhangup = 0;
|
2005-05-15 04:48:30 +00:00
|
|
|
c0->_bridge = c1;
|
|
|
|
c1->_bridge = c0;
|
|
|
|
}
|
2005-03-31 03:00:37 +00:00
|
|
|
continue;
|
|
|
|
}
|
|
|
|
f = ast_read(who);
|
|
|
|
if (!f) {
|
|
|
|
*fo = NULL;
|
|
|
|
*rc = who;
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Didn't get a frame from channel: %s\n",who->name);
|
2005-03-31 03:00:37 +00:00
|
|
|
break;
|
|
|
|
}
|
|
|
|
|
2006-04-16 15:51:53 +00:00
|
|
|
other = (who == c0) ? c1 : c0; /* the 'other' channel */
|
2006-05-31 16:56:50 +00:00
|
|
|
/* Try add the frame info the who's bridged channel jitterbuff */
|
|
|
|
if (jb_in_use)
|
|
|
|
frame_put_in_jb = !ast_jb_put(other, f);
|
2006-04-16 15:51:53 +00:00
|
|
|
|
2005-03-31 03:00:37 +00:00
|
|
|
if ((f->frametype == AST_FRAME_CONTROL) && !(config->flags & AST_BRIDGE_IGNORE_SIGS)) {
|
2006-05-11 10:37:39 +00:00
|
|
|
int bridge_exit = 0;
|
|
|
|
|
2006-05-10 12:24:11 +00:00
|
|
|
switch (f->subclass) {
|
|
|
|
case AST_CONTROL_HOLD:
|
|
|
|
case AST_CONTROL_UNHOLD:
|
|
|
|
case AST_CONTROL_VIDUPDATE:
|
2008-03-05 22:43:22 +00:00
|
|
|
case AST_CONTROL_SRCUPDATE:
|
2008-05-22 16:29:54 +00:00
|
|
|
ast_indicate_data(other, f->subclass, f->data.ptr, f->datalen);
|
2008-08-06 00:30:53 +00:00
|
|
|
if (jb_in_use) {
|
|
|
|
ast_jb_empty_and_reset(c0, c1);
|
|
|
|
}
|
2006-05-10 12:24:11 +00:00
|
|
|
break;
|
|
|
|
default:
|
2005-03-31 03:00:37 +00:00
|
|
|
*fo = f;
|
|
|
|
*rc = who;
|
2006-05-11 10:37:39 +00:00
|
|
|
bridge_exit = 1;
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Got a FRAME_CONTROL (%d) frame on channel %s\n", f->subclass, who->name);
|
2005-03-31 03:00:37 +00:00
|
|
|
break;
|
|
|
|
}
|
2006-05-11 10:37:39 +00:00
|
|
|
if (bridge_exit)
|
2006-05-10 12:24:11 +00:00
|
|
|
break;
|
2005-03-31 03:00:37 +00:00
|
|
|
}
|
|
|
|
if ((f->frametype == AST_FRAME_VOICE) ||
|
2006-08-31 01:59:02 +00:00
|
|
|
(f->frametype == AST_FRAME_DTMF_BEGIN) ||
|
2005-08-09 01:59:59 +00:00
|
|
|
(f->frametype == AST_FRAME_DTMF) ||
|
2006-04-06 15:40:12 +00:00
|
|
|
(f->frametype == AST_FRAME_VIDEO) ||
|
2005-08-09 01:59:59 +00:00
|
|
|
(f->frametype == AST_FRAME_IMAGE) ||
|
|
|
|
(f->frametype == AST_FRAME_HTML) ||
|
2006-01-12 17:14:58 +00:00
|
|
|
(f->frametype == AST_FRAME_MODEM) ||
|
2005-08-09 01:59:59 +00:00
|
|
|
(f->frametype == AST_FRAME_TEXT)) {
|
2006-04-16 15:51:53 +00:00
|
|
|
/* monitored dtmf causes exit from bridge */
|
|
|
|
int monitored_source = (who == c0) ? watch_c0_dtmf : watch_c1_dtmf;
|
|
|
|
|
2006-08-31 01:59:02 +00:00
|
|
|
if (monitored_source &&
|
|
|
|
(f->frametype == AST_FRAME_DTMF_END ||
|
|
|
|
f->frametype == AST_FRAME_DTMF_BEGIN)) {
|
2006-04-16 15:51:53 +00:00
|
|
|
*fo = f;
|
|
|
|
*rc = who;
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Got DTMF %s on channel (%s)\n",
|
|
|
|
f->frametype == AST_FRAME_DTMF_END ? "end" : "begin",
|
|
|
|
who->name);
|
2006-04-16 15:51:53 +00:00
|
|
|
break;
|
2005-03-31 03:00:37 +00:00
|
|
|
}
|
2006-05-31 16:56:50 +00:00
|
|
|
/* Write immediately frames, not passed through jb */
|
|
|
|
if (!frame_put_in_jb)
|
|
|
|
ast_write(other, f);
|
|
|
|
|
|
|
|
/* Check if we have to deliver now */
|
|
|
|
if (jb_in_use)
|
|
|
|
ast_jb_get_and_deliver(c0, c1);
|
2005-03-31 03:00:37 +00:00
|
|
|
}
|
2006-04-16 15:51:53 +00:00
|
|
|
/* XXX do we want to pass on also frames not matched above ? */
|
2005-03-31 03:00:37 +00:00
|
|
|
ast_frfree(f);
|
|
|
|
|
2007-08-08 21:44:58 +00:00
|
|
|
#ifndef HAVE_EPOLL
|
2005-03-31 03:00:37 +00:00
|
|
|
/* Swap who gets priority */
|
|
|
|
cs[2] = cs[0];
|
|
|
|
cs[0] = cs[1];
|
|
|
|
cs[1] = cs[2];
|
2007-08-08 21:44:58 +00:00
|
|
|
#endif
|
2005-03-31 03:00:37 +00:00
|
|
|
}
|
2007-08-08 21:44:58 +00:00
|
|
|
|
|
|
|
ast_poll_channel_del(c0, c1);
|
|
|
|
|
2005-03-31 03:00:37 +00:00
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
2006-09-21 19:27:26 +00:00
|
|
|
/*! \brief Bridge two channels together (early) */
|
|
|
|
int ast_channel_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
|
|
|
|
{
|
|
|
|
/* Make sure we can early bridge, if not error out */
|
|
|
|
if (!c0->tech->early_bridge || (c1 && (!c1->tech->early_bridge || c0->tech->early_bridge != c1->tech->early_bridge)))
|
|
|
|
return -1;
|
|
|
|
|
|
|
|
return c0->tech->early_bridge(c0, c1);
|
|
|
|
}
|
|
|
|
|
2007-12-06 15:29:05 +00:00
|
|
|
/*! \brief Send manager event for bridge link and unlink events.
|
2008-02-15 17:29:08 +00:00
|
|
|
* \param onoff Link/Unlinked
|
|
|
|
* \param type 1 for core, 2 for native
|
|
|
|
* \param c0 first channel in bridge
|
|
|
|
* \param c1 second channel in bridge
|
2007-12-06 15:29:05 +00:00
|
|
|
*/
|
|
|
|
static void manager_bridge_event(int onoff, int type, struct ast_channel *c0, struct ast_channel *c1)
|
|
|
|
{
|
|
|
|
manager_event(EVENT_FLAG_CALL, "Bridge",
|
|
|
|
"Bridgestate: %s\r\n"
|
|
|
|
"Bridgetype: %s\r\n"
|
|
|
|
"Channel1: %s\r\n"
|
|
|
|
"Channel2: %s\r\n"
|
|
|
|
"Uniqueid1: %s\r\n"
|
|
|
|
"Uniqueid2: %s\r\n"
|
|
|
|
"CallerID1: %s\r\n"
|
|
|
|
"CallerID2: %s\r\n",
|
|
|
|
onoff ? "Link" : "Unlink",
|
|
|
|
type == 1 ? "core" : "native",
|
|
|
|
c0->name, c1->name, c0->uniqueid, c1->uniqueid,
|
|
|
|
S_OR(c0->cid.cid_num, ""),
|
|
|
|
S_OR(c1->cid.cid_num, ""));
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Bridge two channels together */
|
2005-08-09 01:59:59 +00:00
|
|
|
enum ast_bridge_result ast_channel_bridge(struct ast_channel *c0, struct ast_channel *c1,
|
2006-04-06 15:40:12 +00:00
|
|
|
struct ast_bridge_config *config, struct ast_frame **fo, struct ast_channel **rc)
|
2005-03-31 03:00:37 +00:00
|
|
|
{
|
|
|
|
struct ast_channel *who = NULL;
|
2005-08-09 01:59:59 +00:00
|
|
|
enum ast_bridge_result res = AST_BRIDGE_COMPLETE;
|
2001-09-19 14:40:16 +00:00
|
|
|
int nativefailed=0;
|
2004-08-06 14:43:25 +00:00
|
|
|
int firstpass;
|
2004-09-04 01:06:42 +00:00
|
|
|
int o0nativeformats;
|
|
|
|
int o1nativeformats;
|
2005-10-13 18:27:54 +00:00
|
|
|
long time_left_ms=0;
|
|
|
|
struct timeval nexteventts = { 0, };
|
2005-08-09 01:59:59 +00:00
|
|
|
char caller_warning = 0;
|
|
|
|
char callee_warning = 0;
|
2004-04-26 23:22:34 +00:00
|
|
|
|
2004-10-23 12:19:47 +00:00
|
|
|
if (c0->_bridge) {
|
2006-04-06 15:40:12 +00:00
|
|
|
ast_log(LOG_WARNING, "%s is already in a bridge with %s\n",
|
2004-10-23 12:19:47 +00:00
|
|
|
c0->name, c0->_bridge->name);
|
2001-05-07 00:43:32 +00:00
|
|
|
return -1;
|
|
|
|
}
|
2004-10-23 12:19:47 +00:00
|
|
|
if (c1->_bridge) {
|
2006-04-06 15:40:12 +00:00
|
|
|
ast_log(LOG_WARNING, "%s is already in a bridge with %s\n",
|
2004-10-23 12:19:47 +00:00
|
|
|
c1->name, c1->_bridge->name);
|
2001-05-07 00:43:32 +00:00
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
2005-08-09 01:59:59 +00:00
|
|
|
/* Stop if we're a zombie or need a soft hangup */
|
|
|
|
if (ast_test_flag(c0, AST_FLAG_ZOMBIE) || ast_check_hangup_locked(c0) ||
|
2006-04-06 15:40:12 +00:00
|
|
|
ast_test_flag(c1, AST_FLAG_ZOMBIE) || ast_check_hangup_locked(c1))
|
2005-08-09 01:59:59 +00:00
|
|
|
return -1;
|
|
|
|
|
|
|
|
*fo = NULL;
|
|
|
|
firstpass = config->firstpass;
|
|
|
|
config->firstpass = 0;
|
|
|
|
|
|
|
|
if (ast_tvzero(config->start_time))
|
|
|
|
config->start_time = ast_tvnow();
|
|
|
|
time_left_ms = config->timelimit;
|
|
|
|
|
|
|
|
caller_warning = ast_test_flag(&config->features_caller, AST_FEATURE_PLAY_WARNING);
|
|
|
|
callee_warning = ast_test_flag(&config->features_callee, AST_FEATURE_PLAY_WARNING);
|
|
|
|
|
|
|
|
if (config->start_sound && firstpass) {
|
|
|
|
if (caller_warning)
|
|
|
|
bridge_playfile(c0, c1, config->start_sound, time_left_ms / 1000);
|
|
|
|
if (callee_warning)
|
|
|
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bridge_playfile(c1, c0, config->start_sound, time_left_ms / 1000);
|
|
|
|
}
|
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
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/* Keep track of bridge */
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2004-10-23 12:19:47 +00:00
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|
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c0->_bridge = c1;
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|
|
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c1->_bridge = c0;
|
2006-04-21 10:05:45 +00:00
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|
|
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2006-04-06 15:40:12 +00:00
|
|
|
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2004-09-04 01:06:42 +00:00
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o0nativeformats = c0->nativeformats;
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2005-08-09 01:59:59 +00:00
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o1nativeformats = c1->nativeformats;
|
|
|
|
|
2006-08-01 23:09:28 +00:00
|
|
|
if (config->feature_timer) {
|
|
|
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nexteventts = ast_tvadd(config->start_time, ast_samp2tv(config->feature_timer, 1000));
|
|
|
|
} else if (config->timelimit) {
|
2005-10-13 18:27:54 +00:00
|
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nexteventts = ast_tvadd(config->start_time, ast_samp2tv(config->timelimit, 1000));
|
|
|
|
if (caller_warning || callee_warning)
|
|
|
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nexteventts = ast_tvsub(nexteventts, ast_samp2tv(config->play_warning, 1000));
|
|
|
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}
|
|
|
|
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
if (!c0->tech->send_digit_begin)
|
|
|
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ast_set_flag(c1, AST_FLAG_END_DTMF_ONLY);
|
|
|
|
if (!c1->tech->send_digit_begin)
|
|
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ast_set_flag(c0, AST_FLAG_END_DTMF_ONLY);
|
2007-12-06 15:29:05 +00:00
|
|
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manager_bridge_event(1, 1, c0, c1);
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
|
2008-03-05 22:43:22 +00:00
|
|
|
/* Before we enter in and bridge these two together tell them both the source of audio has changed */
|
|
|
|
ast_indicate(c0, AST_CONTROL_SRCUPDATE);
|
|
|
|
ast_indicate(c1, AST_CONTROL_SRCUPDATE);
|
|
|
|
|
2001-03-12 03:04:51 +00:00
|
|
|
for (/* ever */;;) {
|
2006-08-05 05:26:29 +00:00
|
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|
struct timeval now = { 0, };
|
2006-08-01 23:09:28 +00:00
|
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|
int to;
|
2008-03-25 15:18:41 +00:00
|
|
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const char *bridge_play_sound = NULL;
|
2006-08-01 23:09:28 +00:00
|
|
|
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2005-10-13 18:27:54 +00:00
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to = -1;
|
2006-08-01 23:09:28 +00:00
|
|
|
|
|
|
|
if (!ast_tvzero(nexteventts)) {
|
2005-10-13 18:27:54 +00:00
|
|
|
now = ast_tvnow();
|
|
|
|
to = ast_tvdiff_ms(nexteventts, now);
|
Merged revisions 43779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r43779 | russell | 2006-09-27 12:55:49 -0400 (Wed, 27 Sep 2006) | 50 lines
Merged revisions 43778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27 Sep 2006) | 42 lines
Fix a problem that occurred if a user entered a digit that matched a bridge
feature that was configured using multiple digits, and the digit that was
pressed timed out in the feature digit timeout period. For example, if blind
transfer is configured as '##', and a user presses just '#'. In this situation,
the call would lock up and no longer pass any frames.
(issue #7977 reported by festr, and issue #7982 reported by michaels and
valuable input provided by mneuhauser and kuj. Fixed by me, with testing help
and peer review from Joshua Colp).
There are a couple of issues involved in this fix:
1) When ast_generic_bridge determines that there has been a timeout, it returned
AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets this result, it calls
ast_generic_bridge over again with the same timestamp for the next event.
This results in an endless loop of nothing until the call is terminated.
This is resolved by simply changing ast_generic_bridge to return
AST_BRIDGE_COMPLETE when it sees a timeout.
2) I also changed ast_channel_bridge such that if in the process of calculating
the time until the next event, it knows a timeout has already occured, to
immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the
channels anyway.
3) In the process of testing the previous two changes, I ran into a problem in
res_features where ast_channel_bridge would return because it determined
that there was a timeout. However, ast_bridge_call in res_features would
then determine by its own calculation that there was still 1 ms before the
timeout really occurs. It would then proceed, and since the bridge broke
out and did *not* return a frame, it interpreted this as the call was over
and hung up the channels.
The reason for this was because ast_bridge_call in res_features and
ast_channel_bridge in channel.c were using different times for their
calculations. channel.c uses the start_time on the bridge config, which
is the time that the feature digit was recieved. However, res_features
had another time, 'start', which was set right before calling
ast_channel_bridge. 'start' will always be slightly after start_time in the
bridge config, and sometimes enough to round up to one ms.
This is fixed by making ast_bridge_call use the same time as
ast_channel_bridge for the timeout calculation.
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43780 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-27 16:57:44 +00:00
|
|
|
if (to <= 0) {
|
2006-11-15 21:36:13 +00:00
|
|
|
if (!config->timelimit) {
|
|
|
|
res = AST_BRIDGE_COMPLETE;
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
to = 0;
|
Merged revisions 43779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r43779 | russell | 2006-09-27 12:55:49 -0400 (Wed, 27 Sep 2006) | 50 lines
Merged revisions 43778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27 Sep 2006) | 42 lines
Fix a problem that occurred if a user entered a digit that matched a bridge
feature that was configured using multiple digits, and the digit that was
pressed timed out in the feature digit timeout period. For example, if blind
transfer is configured as '##', and a user presses just '#'. In this situation,
the call would lock up and no longer pass any frames.
(issue #7977 reported by festr, and issue #7982 reported by michaels and
valuable input provided by mneuhauser and kuj. Fixed by me, with testing help
and peer review from Joshua Colp).
There are a couple of issues involved in this fix:
1) When ast_generic_bridge determines that there has been a timeout, it returned
AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets this result, it calls
ast_generic_bridge over again with the same timestamp for the next event.
This results in an endless loop of nothing until the call is terminated.
This is resolved by simply changing ast_generic_bridge to return
AST_BRIDGE_COMPLETE when it sees a timeout.
2) I also changed ast_channel_bridge such that if in the process of calculating
the time until the next event, it knows a timeout has already occured, to
immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the
channels anyway.
3) In the process of testing the previous two changes, I ran into a problem in
res_features where ast_channel_bridge would return because it determined
that there was a timeout. However, ast_bridge_call in res_features would
then determine by its own calculation that there was still 1 ms before the
timeout really occurs. It would then proceed, and since the bridge broke
out and did *not* return a frame, it interpreted this as the call was over
and hung up the channels.
The reason for this was because ast_bridge_call in res_features and
ast_channel_bridge in channel.c were using different times for their
calculations. channel.c uses the start_time on the bridge config, which
is the time that the feature digit was recieved. However, res_features
had another time, 'start', which was set right before calling
ast_channel_bridge. 'start' will always be slightly after start_time in the
bridge config, and sometimes enough to round up to one ms.
This is fixed by making ast_bridge_call use the same time as
ast_channel_bridge for the timeout calculation.
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43780 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-27 16:57:44 +00:00
|
|
|
}
|
2006-08-01 23:09:28 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
if (config->timelimit) {
|
2005-10-13 18:27:54 +00:00
|
|
|
time_left_ms = config->timelimit - ast_tvdiff_ms(now, config->start_time);
|
|
|
|
if (time_left_ms < to)
|
|
|
|
to = time_left_ms;
|
|
|
|
|
2006-04-18 14:43:32 +00:00
|
|
|
if (time_left_ms <= 0) {
|
|
|
|
if (caller_warning && config->end_sound)
|
2005-08-09 01:59:59 +00:00
|
|
|
bridge_playfile(c0, c1, config->end_sound, 0);
|
2006-04-18 14:43:32 +00:00
|
|
|
if (callee_warning && config->end_sound)
|
2005-08-09 01:59:59 +00:00
|
|
|
bridge_playfile(c1, c0, config->end_sound, 0);
|
2004-04-27 16:42:43 +00:00
|
|
|
*fo = NULL;
|
2006-04-06 15:40:12 +00:00
|
|
|
if (who)
|
2005-05-15 04:48:30 +00:00
|
|
|
*rc = who;
|
2004-04-27 16:42:43 +00:00
|
|
|
res = 0;
|
2004-04-26 23:22:34 +00:00
|
|
|
break;
|
|
|
|
}
|
2005-10-13 18:27:54 +00:00
|
|
|
|
|
|
|
if (!to) {
|
2006-04-16 16:04:24 +00:00
|
|
|
if (time_left_ms >= 5000 && config->warning_sound && config->play_warning) {
|
|
|
|
int t = (time_left_ms + 500) / 1000; /* round to nearest second */
|
|
|
|
if (caller_warning)
|
|
|
|
bridge_playfile(c0, c1, config->warning_sound, t);
|
|
|
|
if (callee_warning)
|
|
|
|
bridge_playfile(c1, c0, config->warning_sound, t);
|
2005-10-13 18:27:54 +00:00
|
|
|
}
|
2007-01-05 22:18:03 +00:00
|
|
|
if (config->warning_freq && (time_left_ms > (config->warning_freq + 5000)))
|
|
|
|
nexteventts = ast_tvadd(nexteventts, ast_samp2tv(config->warning_freq, 1000));
|
|
|
|
else
|
2005-10-13 18:27:54 +00:00
|
|
|
nexteventts = ast_tvadd(config->start_time, ast_samp2tv(config->timelimit, 1000));
|
2004-04-26 23:22:34 +00:00
|
|
|
}
|
|
|
|
}
|
2005-03-23 21:52:31 +00:00
|
|
|
|
|
|
|
if (c0->_softhangup == AST_SOFTHANGUP_UNBRIDGE || c1->_softhangup == AST_SOFTHANGUP_UNBRIDGE) {
|
|
|
|
if (c0->_softhangup == AST_SOFTHANGUP_UNBRIDGE)
|
|
|
|
c0->_softhangup = 0;
|
|
|
|
if (c1->_softhangup == AST_SOFTHANGUP_UNBRIDGE)
|
|
|
|
c1->_softhangup = 0;
|
|
|
|
c0->_bridge = c1;
|
|
|
|
c1->_bridge = c0;
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Unbridge signal received. Ending native bridge.\n");
|
2005-03-23 21:52:31 +00:00
|
|
|
continue;
|
|
|
|
}
|
|
|
|
|
2001-09-19 14:40:16 +00:00
|
|
|
/* Stop if we're a zombie or need a soft hangup */
|
2005-08-09 01:59:59 +00:00
|
|
|
if (ast_test_flag(c0, AST_FLAG_ZOMBIE) || ast_check_hangup_locked(c0) ||
|
|
|
|
ast_test_flag(c1, AST_FLAG_ZOMBIE) || ast_check_hangup_locked(c1)) {
|
2001-09-19 14:40:16 +00:00
|
|
|
*fo = NULL;
|
2005-05-15 04:48:30 +00:00
|
|
|
if (who)
|
|
|
|
*rc = who;
|
2001-09-19 14:40:16 +00:00
|
|
|
res = 0;
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Bridge stops because we're zombie or need a soft hangup: c0=%s, c1=%s, flags: %s,%s,%s,%s\n",
|
|
|
|
c0->name, c1->name,
|
|
|
|
ast_test_flag(c0, AST_FLAG_ZOMBIE) ? "Yes" : "No",
|
|
|
|
ast_check_hangup(c0) ? "Yes" : "No",
|
|
|
|
ast_test_flag(c1, AST_FLAG_ZOMBIE) ? "Yes" : "No",
|
|
|
|
ast_check_hangup(c1) ? "Yes" : "No");
|
2001-09-19 14:40:16 +00:00
|
|
|
break;
|
|
|
|
}
|
2007-06-19 16:25:57 +00:00
|
|
|
|
|
|
|
/* See if the BRIDGEPEER variable needs to be updated */
|
|
|
|
if (!ast_strlen_zero(pbx_builtin_getvar_helper(c0, "BRIDGEPEER")))
|
|
|
|
pbx_builtin_setvar_helper(c0, "BRIDGEPEER", c1->name);
|
|
|
|
if (!ast_strlen_zero(pbx_builtin_getvar_helper(c1, "BRIDGEPEER")))
|
|
|
|
pbx_builtin_setvar_helper(c1, "BRIDGEPEER", c0->name);
|
2008-01-22 20:35:10 +00:00
|
|
|
if (c0->tech->get_pvt_uniqueid)
|
|
|
|
pbx_builtin_setvar_helper(c1, "BRIDGEPVTCALLID", c0->tech->get_pvt_uniqueid(c0));
|
|
|
|
if (c1->tech->get_pvt_uniqueid)
|
|
|
|
pbx_builtin_setvar_helper(c0, "BRIDGEPVTCALLID", c1->tech->get_pvt_uniqueid(c1));
|
2008-03-25 15:18:41 +00:00
|
|
|
|
|
|
|
/* See if we need to play an audio file to any side of the bridge */
|
|
|
|
if ((bridge_play_sound = pbx_builtin_getvar_helper(c0, "BRIDGE_PLAY_SOUND"))) {
|
|
|
|
bridge_playfile(c0, c1, bridge_play_sound, 0);
|
|
|
|
pbx_builtin_setvar_helper(c0, "BRIDGE_PLAY_SOUND", NULL);
|
|
|
|
}
|
|
|
|
if ((bridge_play_sound = pbx_builtin_getvar_helper(c1, "BRIDGE_PLAY_SOUND"))) {
|
|
|
|
bridge_playfile(c1, c0, bridge_play_sound, 0);
|
|
|
|
pbx_builtin_setvar_helper(c1, "BRIDGE_PLAY_SOUND", NULL);
|
|
|
|
}
|
2007-06-19 16:25:57 +00:00
|
|
|
|
2005-08-09 01:59:59 +00:00
|
|
|
if (c0->tech->bridge &&
|
|
|
|
(c0->tech->bridge == c1->tech->bridge) &&
|
2005-10-28 23:01:13 +00:00
|
|
|
!nativefailed && !c0->monitor && !c1->monitor &&
|
2007-08-08 19:30:52 +00:00
|
|
|
!c0->audiohooks && !c1->audiohooks &&
|
2007-10-08 15:05:05 +00:00
|
|
|
!c0->masq && !c0->masqr && !c1->masq && !c1->masqr) {
|
2005-10-28 23:01:13 +00:00
|
|
|
/* Looks like they share a bridge method and nothing else is in the way */
|
2005-03-23 21:52:31 +00:00
|
|
|
ast_set_flag(c0, AST_FLAG_NBRIDGE);
|
|
|
|
ast_set_flag(c1, AST_FLAG_NBRIDGE);
|
2005-10-13 18:27:54 +00:00
|
|
|
if ((res = c0->tech->bridge(c0, c1, config->flags, fo, rc, to)) == AST_BRIDGE_COMPLETE) {
|
2006-04-21 10:05:45 +00:00
|
|
|
/* \todo XXX here should check that cid_num is not NULL */
|
2006-04-06 15:40:12 +00:00
|
|
|
manager_event(EVENT_FLAG_CALL, "Unlink",
|
2005-08-09 01:59:59 +00:00
|
|
|
"Channel1: %s\r\n"
|
|
|
|
"Channel2: %s\r\n"
|
|
|
|
"Uniqueid1: %s\r\n"
|
|
|
|
"Uniqueid2: %s\r\n"
|
|
|
|
"CallerID1: %s\r\n"
|
|
|
|
"CallerID2: %s\r\n",
|
|
|
|
c0->name, c1->name, c0->uniqueid, c1->uniqueid, c0->cid.cid_num, c1->cid.cid_num);
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Returning from native bridge, channels: %s, %s\n", c0->name, c1->name);
|
2005-08-09 01:59:59 +00:00
|
|
|
|
2005-03-23 21:52:31 +00:00
|
|
|
ast_clear_flag(c0, AST_FLAG_NBRIDGE);
|
|
|
|
ast_clear_flag(c1, AST_FLAG_NBRIDGE);
|
2005-08-09 01:59:59 +00:00
|
|
|
|
|
|
|
if (c0->_softhangup == AST_SOFTHANGUP_UNBRIDGE || c1->_softhangup == AST_SOFTHANGUP_UNBRIDGE)
|
2005-03-23 21:52:31 +00:00
|
|
|
continue;
|
2005-08-09 01:59:59 +00:00
|
|
|
|
|
|
|
c0->_bridge = NULL;
|
|
|
|
c1->_bridge = NULL;
|
|
|
|
|
|
|
|
return res;
|
2005-03-23 21:52:31 +00:00
|
|
|
} else {
|
|
|
|
ast_clear_flag(c0, AST_FLAG_NBRIDGE);
|
|
|
|
ast_clear_flag(c1, AST_FLAG_NBRIDGE);
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
2005-08-09 01:59:59 +00:00
|
|
|
switch (res) {
|
|
|
|
case AST_BRIDGE_RETRY:
|
2005-11-02 17:20:35 +00:00
|
|
|
continue;
|
2005-08-09 01:59:59 +00:00
|
|
|
default:
|
2007-07-26 15:49:18 +00:00
|
|
|
ast_verb(3, "Native bridging %s and %s ended\n", c0->name, c1->name);
|
2005-08-09 01:59:59 +00:00
|
|
|
/* fallthrough */
|
|
|
|
case AST_BRIDGE_FAILED_NOWARN:
|
2005-07-25 22:13:32 +00:00
|
|
|
nativefailed++;
|
2005-08-09 01:59:59 +00:00
|
|
|
break;
|
|
|
|
}
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
|
|
|
|
2005-08-09 01:59:59 +00:00
|
|
|
if (((c0->writeformat != c1->readformat) || (c0->readformat != c1->writeformat) ||
|
|
|
|
(c0->nativeformats != o0nativeformats) || (c1->nativeformats != o1nativeformats)) &&
|
|
|
|
!(c0->generator || c1->generator)) {
|
2001-09-19 14:40:16 +00:00
|
|
|
if (ast_channel_make_compatible(c0, c1)) {
|
|
|
|
ast_log(LOG_WARNING, "Can't make %s and %s compatible\n", c0->name, c1->name);
|
2007-12-06 15:29:05 +00:00
|
|
|
manager_bridge_event(0, 1, c0, c1);
|
2005-08-09 01:59:59 +00:00
|
|
|
return AST_BRIDGE_FAILED;
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
2004-09-04 01:06:42 +00:00
|
|
|
o0nativeformats = c0->nativeformats;
|
|
|
|
o1nativeformats = c1->nativeformats;
|
2001-09-19 14:40:16 +00:00
|
|
|
}
|
2008-06-10 12:54:39 +00:00
|
|
|
|
|
|
|
if (!ast_strlen_zero(pbx_builtin_getvar_helper(c0, "BRIDGEPEER")))
|
|
|
|
pbx_builtin_setvar_helper(c0, "BRIDGEPEER", c1->name);
|
|
|
|
if (!ast_strlen_zero(pbx_builtin_getvar_helper(c1, "BRIDGEPEER")))
|
|
|
|
pbx_builtin_setvar_helper(c1, "BRIDGEPEER", c0->name);
|
|
|
|
|
2006-01-05 23:52:34 +00:00
|
|
|
res = ast_generic_bridge(c0, c1, config, fo, rc, nexteventts);
|
2005-08-09 01:59:59 +00:00
|
|
|
if (res != AST_BRIDGE_RETRY)
|
2001-05-07 00:43:32 +00:00
|
|
|
break;
|
2001-03-12 03:04:51 +00:00
|
|
|
}
|
2005-08-09 01:59:59 +00:00
|
|
|
|
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
|
|
|
ast_clear_flag(c0, AST_FLAG_END_DTMF_ONLY);
|
|
|
|
ast_clear_flag(c1, AST_FLAG_END_DTMF_ONLY);
|
|
|
|
|
2008-03-05 22:43:22 +00:00
|
|
|
/* Now that we have broken the bridge the source will change yet again */
|
|
|
|
ast_indicate(c0, AST_CONTROL_SRCUPDATE);
|
|
|
|
ast_indicate(c1, AST_CONTROL_SRCUPDATE);
|
|
|
|
|
2004-10-23 12:19:47 +00:00
|
|
|
c0->_bridge = NULL;
|
|
|
|
c1->_bridge = NULL;
|
2005-08-09 01:59:59 +00:00
|
|
|
|
2006-04-21 10:05:45 +00:00
|
|
|
/* \todo XXX here should check that cid_num is not NULL */
|
2005-06-21 23:10:29 +00:00
|
|
|
manager_event(EVENT_FLAG_CALL, "Unlink",
|
2005-08-09 01:59:59 +00:00
|
|
|
"Channel1: %s\r\n"
|
|
|
|
"Channel2: %s\r\n"
|
|
|
|
"Uniqueid1: %s\r\n"
|
|
|
|
"Uniqueid2: %s\r\n"
|
|
|
|
"CallerID1: %s\r\n"
|
|
|
|
"CallerID2: %s\r\n",
|
|
|
|
c0->name, c1->name, c0->uniqueid, c1->uniqueid, c0->cid.cid_num, c1->cid.cid_num);
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Bridge stops bridging channels %s and %s\n", c0->name, c1->name);
|
2005-08-09 01:59:59 +00:00
|
|
|
|
2001-05-07 00:43:32 +00:00
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Sets an option on a channel */
|
2001-05-07 00:43:32 +00:00
|
|
|
int ast_channel_setoption(struct ast_channel *chan, int option, void *data, int datalen, int block)
|
|
|
|
{
|
2008-02-18 23:47:01 +00:00
|
|
|
if (!chan->tech->setoption) {
|
2001-05-07 00:43:32 +00:00
|
|
|
errno = ENOSYS;
|
|
|
|
return -1;
|
|
|
|
}
|
2008-02-18 23:47:01 +00:00
|
|
|
|
|
|
|
if (block)
|
2001-05-07 00:43:32 +00:00
|
|
|
ast_log(LOG_ERROR, "XXX Blocking not implemented yet XXX\n");
|
2008-02-18 23:47:01 +00:00
|
|
|
|
|
|
|
return chan->tech->setoption(chan, option, data, datalen);
|
|
|
|
}
|
|
|
|
|
|
|
|
int ast_channel_queryoption(struct ast_channel *chan, int option, void *data, int *datalen, int block)
|
|
|
|
{
|
|
|
|
if (!chan->tech->queryoption) {
|
|
|
|
errno = ENOSYS;
|
2001-05-07 00:43:32 +00:00
|
|
|
return -1;
|
|
|
|
}
|
2008-02-18 23:47:01 +00:00
|
|
|
|
|
|
|
if (block)
|
|
|
|
ast_log(LOG_ERROR, "XXX Blocking not implemented yet XXX\n");
|
|
|
|
|
|
|
|
return chan->tech->queryoption(chan, option, data, datalen);
|
2001-03-12 03:04:51 +00:00
|
|
|
}
|
2001-12-20 15:21:47 +00:00
|
|
|
|
2002-06-21 01:40:13 +00:00
|
|
|
struct tonepair_def {
|
|
|
|
int freq1;
|
|
|
|
int freq2;
|
|
|
|
int duration;
|
|
|
|
int vol;
|
|
|
|
};
|
|
|
|
|
|
|
|
struct tonepair_state {
|
2006-08-15 20:25:13 +00:00
|
|
|
int fac1;
|
|
|
|
int fac2;
|
|
|
|
int v1_1;
|
|
|
|
int v2_1;
|
|
|
|
int v3_1;
|
|
|
|
int v1_2;
|
|
|
|
int v2_2;
|
|
|
|
int v3_2;
|
2002-06-21 01:40:13 +00:00
|
|
|
int origwfmt;
|
2006-08-15 20:25:13 +00:00
|
|
|
int pos;
|
|
|
|
int duration;
|
|
|
|
int modulate;
|
2002-06-21 01:40:13 +00:00
|
|
|
struct ast_frame f;
|
|
|
|
unsigned char offset[AST_FRIENDLY_OFFSET];
|
|
|
|
short data[4000];
|
|
|
|
};
|
|
|
|
|
|
|
|
static void tonepair_release(struct ast_channel *chan, void *params)
|
|
|
|
{
|
|
|
|
struct tonepair_state *ts = params;
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2006-04-14 20:47:06 +00:00
|
|
|
if (chan)
|
2004-04-06 22:17:32 +00:00
|
|
|
ast_set_write_format(chan, ts->origwfmt);
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(ts);
|
2002-06-21 01:40:13 +00:00
|
|
|
}
|
|
|
|
|
2005-07-25 22:13:32 +00:00
|
|
|
static void *tonepair_alloc(struct ast_channel *chan, void *params)
|
2002-06-21 01:40:13 +00:00
|
|
|
{
|
|
|
|
struct tonepair_state *ts;
|
|
|
|
struct tonepair_def *td = params;
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2006-02-15 01:48:54 +00:00
|
|
|
if (!(ts = ast_calloc(1, sizeof(*ts))))
|
2002-06-21 01:40:13 +00:00
|
|
|
return NULL;
|
|
|
|
ts->origwfmt = chan->writeformat;
|
2004-04-06 22:17:32 +00:00
|
|
|
if (ast_set_write_format(chan, AST_FORMAT_SLINEAR)) {
|
2002-06-21 01:40:13 +00:00
|
|
|
ast_log(LOG_WARNING, "Unable to set '%s' to signed linear format (write)\n", chan->name);
|
|
|
|
tonepair_release(NULL, ts);
|
|
|
|
ts = NULL;
|
|
|
|
} else {
|
2006-08-15 20:25:13 +00:00
|
|
|
ts->fac1 = 2.0 * cos(2.0 * M_PI * (td->freq1 / 8000.0)) * 32768.0;
|
|
|
|
ts->v1_1 = 0;
|
|
|
|
ts->v2_1 = sin(-4.0 * M_PI * (td->freq1 / 8000.0)) * td->vol;
|
|
|
|
ts->v3_1 = sin(-2.0 * M_PI * (td->freq1 / 8000.0)) * td->vol;
|
|
|
|
ts->v2_1 = 0;
|
|
|
|
ts->fac2 = 2.0 * cos(2.0 * M_PI * (td->freq2 / 8000.0)) * 32768.0;
|
|
|
|
ts->v2_2 = sin(-4.0 * M_PI * (td->freq2 / 8000.0)) * td->vol;
|
|
|
|
ts->v3_2 = sin(-2.0 * M_PI * (td->freq2 / 8000.0)) * td->vol;
|
2002-06-21 01:40:13 +00:00
|
|
|
ts->duration = td->duration;
|
2006-08-15 20:25:13 +00:00
|
|
|
ts->modulate = 0;
|
2002-06-21 01:40:13 +00:00
|
|
|
}
|
|
|
|
/* Let interrupts interrupt :) */
|
2004-12-07 20:38:43 +00:00
|
|
|
ast_set_flag(chan, AST_FLAG_WRITE_INT);
|
2002-06-21 01:40:13 +00:00
|
|
|
return ts;
|
|
|
|
}
|
|
|
|
|
2003-02-05 21:18:09 +00:00
|
|
|
static int tonepair_generator(struct ast_channel *chan, void *data, int len, int samples)
|
2002-06-21 01:40:13 +00:00
|
|
|
{
|
|
|
|
struct tonepair_state *ts = data;
|
|
|
|
int x;
|
2003-02-05 21:18:09 +00:00
|
|
|
|
2006-04-06 15:40:12 +00:00
|
|
|
/* we need to prepare a frame with 16 * timelen samples as we're
|
2003-02-05 21:18:09 +00:00
|
|
|
* generating SLIN audio
|
|
|
|
*/
|
|
|
|
len = samples * 2;
|
|
|
|
|
2002-06-21 01:40:13 +00:00
|
|
|
if (len > sizeof(ts->data) / 2 - 1) {
|
|
|
|
ast_log(LOG_WARNING, "Can't generate that much data!\n");
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
memset(&ts->f, 0, sizeof(ts->f));
|
2006-08-15 20:25:13 +00:00
|
|
|
for (x=0;x<len/2;x++) {
|
|
|
|
ts->v1_1 = ts->v2_1;
|
|
|
|
ts->v2_1 = ts->v3_1;
|
|
|
|
ts->v3_1 = (ts->fac1 * ts->v2_1 >> 15) - ts->v1_1;
|
|
|
|
|
|
|
|
ts->v1_2 = ts->v2_2;
|
|
|
|
ts->v2_2 = ts->v3_2;
|
|
|
|
ts->v3_2 = (ts->fac2 * ts->v2_2 >> 15) - ts->v1_2;
|
|
|
|
if (ts->modulate) {
|
|
|
|
int p;
|
|
|
|
p = ts->v3_2 - 32768;
|
|
|
|
if (p < 0) p = -p;
|
|
|
|
p = ((p * 9) / 10) + 1;
|
|
|
|
ts->data[x] = (ts->v3_1 * p) >> 15;
|
|
|
|
} else
|
|
|
|
ts->data[x] = ts->v3_1 + ts->v3_2;
|
|
|
|
}
|
2002-06-21 01:40:13 +00:00
|
|
|
ts->f.frametype = AST_FRAME_VOICE;
|
|
|
|
ts->f.subclass = AST_FORMAT_SLINEAR;
|
|
|
|
ts->f.datalen = len;
|
2003-02-05 21:18:09 +00:00
|
|
|
ts->f.samples = samples;
|
2002-06-21 01:40:13 +00:00
|
|
|
ts->f.offset = AST_FRIENDLY_OFFSET;
|
2008-05-22 16:29:54 +00:00
|
|
|
ts->f.data.ptr = ts->data;
|
2002-06-21 01:40:13 +00:00
|
|
|
ast_write(chan, &ts->f);
|
|
|
|
ts->pos += x;
|
|
|
|
if (ts->duration > 0) {
|
|
|
|
if (ts->pos >= ts->duration * 8)
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static struct ast_generator tonepair = {
|
|
|
|
alloc: tonepair_alloc,
|
|
|
|
release: tonepair_release,
|
|
|
|
generate: tonepair_generator,
|
|
|
|
};
|
|
|
|
|
|
|
|
int ast_tonepair_start(struct ast_channel *chan, int freq1, int freq2, int duration, int vol)
|
|
|
|
{
|
|
|
|
struct tonepair_def d = { 0, };
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2002-06-21 01:40:13 +00:00
|
|
|
d.freq1 = freq1;
|
|
|
|
d.freq2 = freq2;
|
|
|
|
d.duration = duration;
|
2006-04-06 15:40:12 +00:00
|
|
|
d.vol = (vol < 1) ? 8192 : vol; /* force invalid to 8192 */
|
2002-06-21 01:40:13 +00:00
|
|
|
if (ast_activate_generator(chan, &tonepair, &d))
|
|
|
|
return -1;
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
void ast_tonepair_stop(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
ast_deactivate_generator(chan);
|
|
|
|
}
|
|
|
|
|
|
|
|
int ast_tonepair(struct ast_channel *chan, int freq1, int freq2, int duration, int vol)
|
|
|
|
{
|
|
|
|
int res;
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2002-06-21 01:40:13 +00:00
|
|
|
if ((res = ast_tonepair_start(chan, freq1, freq2, duration, vol)))
|
|
|
|
return res;
|
|
|
|
|
|
|
|
/* Give us some wiggle room */
|
2006-04-06 15:40:12 +00:00
|
|
|
while (chan->generatordata && ast_waitfor(chan, 100) >= 0) {
|
|
|
|
struct ast_frame *f = ast_read(chan);
|
2002-06-21 01:40:13 +00:00
|
|
|
if (f)
|
|
|
|
ast_frfree(f);
|
|
|
|
else
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
return 0;
|
|
|
|
}
|
2003-03-25 19:30:06 +00:00
|
|
|
|
2006-11-16 08:18:41 +00:00
|
|
|
ast_group_t ast_get_group(const char *s)
|
2004-02-27 06:15:49 +00:00
|
|
|
{
|
|
|
|
char *piece;
|
2006-05-03 17:44:30 +00:00
|
|
|
char *c;
|
2005-07-25 22:13:32 +00:00
|
|
|
int start=0, finish=0, x;
|
2005-01-15 21:51:38 +00:00
|
|
|
ast_group_t group = 0;
|
2005-07-25 22:13:32 +00:00
|
|
|
|
2007-12-27 21:41:22 +00:00
|
|
|
if (ast_strlen_zero(s))
|
|
|
|
return 0;
|
|
|
|
|
2006-05-03 17:44:30 +00:00
|
|
|
c = ast_strdupa(s);
|
2004-02-27 06:15:49 +00:00
|
|
|
|
2005-12-24 05:18:40 +00:00
|
|
|
while ((piece = strsep(&c, ","))) {
|
2004-02-27 06:15:49 +00:00
|
|
|
if (sscanf(piece, "%d-%d", &start, &finish) == 2) {
|
|
|
|
/* Range */
|
|
|
|
} else if (sscanf(piece, "%d", &start)) {
|
|
|
|
/* Just one */
|
|
|
|
finish = start;
|
|
|
|
} else {
|
2005-07-25 22:13:32 +00:00
|
|
|
ast_log(LOG_ERROR, "Syntax error parsing group configuration '%s' at '%s'. Ignoring.\n", s, piece);
|
2005-01-13 03:31:58 +00:00
|
|
|
continue;
|
2004-02-27 06:15:49 +00:00
|
|
|
}
|
2005-07-25 22:13:32 +00:00
|
|
|
for (x = start; x <= finish; x++) {
|
2005-01-15 21:51:38 +00:00
|
|
|
if ((x > 63) || (x < 0)) {
|
|
|
|
ast_log(LOG_WARNING, "Ignoring invalid group %d (maximum group is 63)\n", x);
|
2004-02-27 06:15:49 +00:00
|
|
|
} else
|
2005-07-12 15:13:39 +00:00
|
|
|
group |= ((ast_group_t) 1 << x);
|
2004-02-27 06:15:49 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
return group;
|
|
|
|
}
|
2004-12-09 19:55:01 +00:00
|
|
|
|
2006-07-19 20:44:39 +00:00
|
|
|
static int (*ast_moh_start_ptr)(struct ast_channel *, const char *, const char *) = NULL;
|
2004-12-09 19:55:01 +00:00
|
|
|
static void (*ast_moh_stop_ptr)(struct ast_channel *) = NULL;
|
2004-12-24 01:40:07 +00:00
|
|
|
static void (*ast_moh_cleanup_ptr)(struct ast_channel *) = NULL;
|
2004-12-09 19:55:01 +00:00
|
|
|
|
2006-07-19 20:44:39 +00:00
|
|
|
void ast_install_music_functions(int (*start_ptr)(struct ast_channel *, const char *, const char *),
|
2006-02-01 23:05:28 +00:00
|
|
|
void (*stop_ptr)(struct ast_channel *),
|
|
|
|
void (*cleanup_ptr)(struct ast_channel *))
|
2004-12-09 19:55:01 +00:00
|
|
|
{
|
|
|
|
ast_moh_start_ptr = start_ptr;
|
|
|
|
ast_moh_stop_ptr = stop_ptr;
|
2004-12-24 01:40:07 +00:00
|
|
|
ast_moh_cleanup_ptr = cleanup_ptr;
|
2004-12-09 19:55:01 +00:00
|
|
|
}
|
|
|
|
|
2006-04-06 15:40:12 +00:00
|
|
|
void ast_uninstall_music_functions(void)
|
2004-12-09 19:55:01 +00:00
|
|
|
{
|
|
|
|
ast_moh_start_ptr = NULL;
|
|
|
|
ast_moh_stop_ptr = NULL;
|
2004-12-24 01:40:07 +00:00
|
|
|
ast_moh_cleanup_ptr = NULL;
|
2004-12-09 19:55:01 +00:00
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Turn on music on hold on a given channel */
|
2006-07-19 20:44:39 +00:00
|
|
|
int ast_moh_start(struct ast_channel *chan, const char *mclass, const char *interpclass)
|
2004-12-09 19:55:01 +00:00
|
|
|
{
|
2005-07-25 22:13:32 +00:00
|
|
|
if (ast_moh_start_ptr)
|
2006-07-19 20:44:39 +00:00
|
|
|
return ast_moh_start_ptr(chan, mclass, interpclass);
|
|
|
|
|
2007-07-26 15:49:18 +00:00
|
|
|
ast_verb(3, "Music class %s requested but no musiconhold loaded.\n", mclass ? mclass : (interpclass ? interpclass : "default"));
|
2004-12-09 19:55:01 +00:00
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Turn off music on hold on a given channel */
|
2006-04-06 15:40:12 +00:00
|
|
|
void ast_moh_stop(struct ast_channel *chan)
|
2004-12-09 19:55:01 +00:00
|
|
|
{
|
2005-12-24 05:18:40 +00:00
|
|
|
if (ast_moh_stop_ptr)
|
2004-12-09 19:55:01 +00:00
|
|
|
ast_moh_stop_ptr(chan);
|
|
|
|
}
|
2004-12-24 01:40:07 +00:00
|
|
|
|
2006-04-06 15:40:12 +00:00
|
|
|
void ast_moh_cleanup(struct ast_channel *chan)
|
2004-12-24 01:40:07 +00:00
|
|
|
{
|
2005-12-24 05:18:40 +00:00
|
|
|
if (ast_moh_cleanup_ptr)
|
|
|
|
ast_moh_cleanup_ptr(chan);
|
2004-12-24 01:40:07 +00:00
|
|
|
}
|
|
|
|
|
2004-12-31 00:04:41 +00:00
|
|
|
void ast_channels_init(void)
|
|
|
|
{
|
2008-01-28 04:30:44 +00:00
|
|
|
ast_cli_register_multiple(cli_channel, ARRAY_LEN(cli_channel));
|
2004-12-31 00:04:41 +00:00
|
|
|
}
|
2005-01-15 21:51:38 +00:00
|
|
|
|
2005-12-31 16:35:48 +00:00
|
|
|
/*! \brief Print call group and pickup group ---*/
|
2006-04-06 15:40:12 +00:00
|
|
|
char *ast_print_group(char *buf, int buflen, ast_group_t group)
|
2005-01-15 21:51:38 +00:00
|
|
|
{
|
|
|
|
unsigned int i;
|
2008-01-28 04:30:44 +00:00
|
|
|
int first = 1;
|
2005-01-15 21:51:38 +00:00
|
|
|
char num[3];
|
|
|
|
|
|
|
|
buf[0] = '\0';
|
|
|
|
|
|
|
|
if (!group) /* Return empty string if no group */
|
2005-12-24 05:18:40 +00:00
|
|
|
return buf;
|
2005-01-15 21:51:38 +00:00
|
|
|
|
2005-12-24 05:18:40 +00:00
|
|
|
for (i = 0; i <= 63; i++) { /* Max group is 63 */
|
2005-07-12 15:13:39 +00:00
|
|
|
if (group & ((ast_group_t) 1 << i)) {
|
2005-01-15 21:51:38 +00:00
|
|
|
if (!first) {
|
2008-03-07 06:54:47 +00:00
|
|
|
strncat(buf, ", ", buflen - strlen(buf) - 1);
|
2005-01-15 21:51:38 +00:00
|
|
|
} else {
|
2008-01-28 04:30:44 +00:00
|
|
|
first = 0;
|
2005-01-15 21:51:38 +00:00
|
|
|
}
|
|
|
|
snprintf(num, sizeof(num), "%u", i);
|
2008-03-07 06:54:47 +00:00
|
|
|
strncat(buf, num, buflen - strlen(buf) - 1);
|
2005-01-15 21:51:38 +00:00
|
|
|
}
|
|
|
|
}
|
2005-12-24 05:18:40 +00:00
|
|
|
return buf;
|
2005-01-15 21:51:38 +00:00
|
|
|
}
|
2005-07-15 23:24:51 +00:00
|
|
|
|
|
|
|
void ast_set_variables(struct ast_channel *chan, struct ast_variable *vars)
|
|
|
|
{
|
|
|
|
struct ast_variable *cur;
|
|
|
|
|
|
|
|
for (cur = vars; cur; cur = cur->next)
|
|
|
|
pbx_builtin_setvar_helper(chan, cur->name, cur->value);
|
|
|
|
}
|
2005-10-28 23:01:13 +00:00
|
|
|
|
2005-11-01 17:22:25 +00:00
|
|
|
static void *silence_generator_alloc(struct ast_channel *chan, void *data)
|
|
|
|
{
|
|
|
|
/* just store the data pointer in the channel structure */
|
|
|
|
return data;
|
|
|
|
}
|
|
|
|
|
|
|
|
static void silence_generator_release(struct ast_channel *chan, void *data)
|
|
|
|
{
|
|
|
|
/* nothing to do */
|
|
|
|
}
|
|
|
|
|
2006-04-06 15:40:12 +00:00
|
|
|
static int silence_generator_generate(struct ast_channel *chan, void *data, int len, int samples)
|
2005-11-01 17:22:25 +00:00
|
|
|
{
|
2006-04-06 15:40:12 +00:00
|
|
|
short buf[samples];
|
|
|
|
struct ast_frame frame = {
|
|
|
|
.frametype = AST_FRAME_VOICE,
|
|
|
|
.subclass = AST_FORMAT_SLINEAR,
|
2008-05-22 16:29:54 +00:00
|
|
|
.data.ptr = buf,
|
2006-04-06 15:40:12 +00:00
|
|
|
.samples = samples,
|
|
|
|
.datalen = sizeof(buf),
|
|
|
|
};
|
2008-01-28 04:30:44 +00:00
|
|
|
|
2006-04-19 06:58:20 +00:00
|
|
|
memset(buf, 0, sizeof(buf));
|
2008-01-28 04:30:44 +00:00
|
|
|
|
2006-04-06 15:40:12 +00:00
|
|
|
if (ast_write(chan, &frame))
|
|
|
|
return -1;
|
2008-01-28 04:30:44 +00:00
|
|
|
|
2005-11-01 17:22:25 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static struct ast_generator silence_generator = {
|
|
|
|
.alloc = silence_generator_alloc,
|
|
|
|
.release = silence_generator_release,
|
2006-04-06 15:40:12 +00:00
|
|
|
.generate = silence_generator_generate,
|
2005-11-01 17:22:25 +00:00
|
|
|
};
|
|
|
|
|
|
|
|
struct ast_silence_generator {
|
|
|
|
int old_write_format;
|
|
|
|
};
|
|
|
|
|
|
|
|
struct ast_silence_generator *ast_channel_start_silence_generator(struct ast_channel *chan)
|
|
|
|
{
|
|
|
|
struct ast_silence_generator *state;
|
|
|
|
|
2006-02-15 01:48:54 +00:00
|
|
|
if (!(state = ast_calloc(1, sizeof(*state)))) {
|
2005-11-01 17:22:25 +00:00
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
state->old_write_format = chan->writeformat;
|
|
|
|
|
|
|
|
if (ast_set_write_format(chan, AST_FORMAT_SLINEAR) < 0) {
|
|
|
|
ast_log(LOG_ERROR, "Could not set write format to SLINEAR\n");
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(state);
|
2005-11-01 17:22:25 +00:00
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
ast_activate_generator(chan, &silence_generator, state);
|
|
|
|
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Started silence generator on '%s'\n", chan->name);
|
2005-11-01 17:22:25 +00:00
|
|
|
|
|
|
|
return state;
|
|
|
|
}
|
|
|
|
|
|
|
|
void ast_channel_stop_silence_generator(struct ast_channel *chan, struct ast_silence_generator *state)
|
|
|
|
{
|
|
|
|
if (!state)
|
|
|
|
return;
|
|
|
|
|
|
|
|
ast_deactivate_generator(chan);
|
|
|
|
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "Stopped silence generator on '%s'\n", chan->name);
|
2005-11-01 17:22:25 +00:00
|
|
|
|
|
|
|
if (ast_set_write_format(chan, state->old_write_format) < 0)
|
|
|
|
ast_log(LOG_ERROR, "Could not return write format to its original state\n");
|
|
|
|
|
2007-06-06 21:20:11 +00:00
|
|
|
ast_free(state);
|
2005-11-01 17:22:25 +00:00
|
|
|
}
|
2006-01-24 12:48:44 +00:00
|
|
|
|
|
|
|
|
|
|
|
/*! \ brief Convert channel reloadreason (ENUM) to text string for manager event */
|
2006-04-06 15:40:12 +00:00
|
|
|
const char *channelreloadreason2txt(enum channelreloadreason reason)
|
|
|
|
{
|
2006-01-24 12:48:44 +00:00
|
|
|
switch (reason) {
|
2006-04-06 15:40:12 +00:00
|
|
|
case CHANNEL_MODULE_LOAD:
|
|
|
|
return "LOAD (Channel module load)";
|
|
|
|
|
|
|
|
case CHANNEL_MODULE_RELOAD:
|
|
|
|
return "RELOAD (Channel module reload)";
|
|
|
|
|
|
|
|
case CHANNEL_CLI_RELOAD:
|
|
|
|
return "CLIRELOAD (Channel module reload by CLI command)";
|
|
|
|
|
|
|
|
default:
|
|
|
|
return "MANAGERRELOAD (Channel module reload by manager)";
|
2006-01-24 12:48:44 +00:00
|
|
|
}
|
|
|
|
};
|
2006-04-15 08:07:50 +00:00
|
|
|
|
|
|
|
#ifdef DEBUG_CHANNEL_LOCKS
|
|
|
|
|
|
|
|
/*! \brief Unlock AST channel (and print debugging output)
|
|
|
|
\note You need to enable DEBUG_CHANNEL_LOCKS for this function
|
|
|
|
*/
|
2008-05-13 23:54:01 +00:00
|
|
|
int __ast_channel_unlock(struct ast_channel *chan, const char *filename, int lineno, const char *func)
|
2006-04-15 08:07:50 +00:00
|
|
|
{
|
|
|
|
int res = 0;
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(3, "::::==== Unlocking AST channel %s\n", chan->name);
|
2006-04-15 08:07:50 +00:00
|
|
|
|
|
|
|
if (!chan) {
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(1, "::::==== Unlocking non-existing channel \n");
|
2006-04-15 08:07:50 +00:00
|
|
|
return 0;
|
|
|
|
}
|
2008-05-13 23:54:01 +00:00
|
|
|
#ifdef DEBUG_THREADS
|
|
|
|
res = __ast_pthread_mutex_unlock(filename, lineno, func, "(channel lock)", &chan->lock_dont_use);
|
|
|
|
#else
|
2007-11-15 16:20:47 +00:00
|
|
|
res = ast_mutex_unlock(&chan->lock_dont_use);
|
2008-05-13 23:54:01 +00:00
|
|
|
#endif
|
2006-04-15 08:07:50 +00:00
|
|
|
|
|
|
|
if (option_debug > 2) {
|
|
|
|
#ifdef DEBUG_THREADS
|
2006-10-18 02:46:39 +00:00
|
|
|
int count = 0;
|
2008-06-03 18:26:51 +00:00
|
|
|
if ((count = chan->lock_dont_use.track.reentrancy))
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(3, ":::=== Still have %d locks (recursive)\n", count);
|
2006-04-15 08:07:50 +00:00
|
|
|
#endif
|
|
|
|
if (!res)
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(3, "::::==== Channel %s was unlocked\n", chan->name);
|
2006-04-15 08:07:50 +00:00
|
|
|
if (res == EINVAL) {
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(3, "::::==== Channel %s had no lock by this thread. Failed unlocking\n", chan->name);
|
2006-04-15 08:07:50 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
if (res == EPERM) {
|
|
|
|
/* We had no lock, so okay any way*/
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(4, "::::==== Channel %s was not locked at all \n", chan->name);
|
2006-04-15 08:07:50 +00:00
|
|
|
res = 0;
|
|
|
|
}
|
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
|
|
|
/*! \brief Lock AST channel (and print debugging output)
|
|
|
|
\note You need to enable DEBUG_CHANNEL_LOCKS for this function */
|
2008-05-13 23:54:01 +00:00
|
|
|
int __ast_channel_lock(struct ast_channel *chan, const char *filename, int lineno, const char *func)
|
2006-04-15 08:07:50 +00:00
|
|
|
{
|
|
|
|
int res;
|
|
|
|
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(4, "====:::: Locking AST channel %s\n", chan->name);
|
2006-04-15 08:07:50 +00:00
|
|
|
|
2008-05-13 23:54:01 +00:00
|
|
|
#ifdef DEBUG_THREADS
|
|
|
|
res = __ast_pthread_mutex_lock(filename, lineno, func, "(channel lock)", &chan->lock_dont_use);
|
|
|
|
#else
|
2007-11-15 16:20:47 +00:00
|
|
|
res = ast_mutex_lock(&chan->lock_dont_use);
|
2008-05-13 23:54:01 +00:00
|
|
|
#endif
|
2006-04-15 08:07:50 +00:00
|
|
|
|
|
|
|
if (option_debug > 3) {
|
|
|
|
#ifdef DEBUG_THREADS
|
2006-10-18 02:46:39 +00:00
|
|
|
int count = 0;
|
2008-06-03 18:26:51 +00:00
|
|
|
if ((count = chan->lock_dont_use.track.reentrancy))
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(4, ":::=== Now have %d locks (recursive)\n", count);
|
2006-04-15 08:07:50 +00:00
|
|
|
#endif
|
|
|
|
if (!res)
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(4, "::::==== Channel %s was locked\n", chan->name);
|
2006-04-15 08:07:50 +00:00
|
|
|
if (res == EDEADLK) {
|
2006-12-01 20:49:06 +00:00
|
|
|
/* We had no lock, so okey any way */
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(4, "::::==== Channel %s was not locked by us. Lock would cause deadlock.\n", chan->name);
|
2006-04-15 08:07:50 +00:00
|
|
|
}
|
|
|
|
if (res == EINVAL) {
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(4, "::::==== Channel %s lock failed. No mutex.\n", chan->name);
|
2006-04-15 08:07:50 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
|
|
|
/*! \brief Lock AST channel (and print debugging output)
|
|
|
|
\note You need to enable DEBUG_CHANNEL_LOCKS for this function */
|
2008-05-13 23:54:01 +00:00
|
|
|
int __ast_channel_trylock(struct ast_channel *chan, const char *filename, int lineno, const char *func)
|
2006-04-15 08:07:50 +00:00
|
|
|
{
|
|
|
|
int res;
|
|
|
|
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(3, "====:::: Trying to lock AST channel %s\n", chan->name);
|
2008-05-13 23:54:01 +00:00
|
|
|
#ifdef DEBUG_THREADS
|
|
|
|
res = __ast_pthread_mutex_trylock(filename, lineno, func, "(channel lock)", &chan->lock_dont_use);
|
|
|
|
#else
|
2007-11-15 16:20:47 +00:00
|
|
|
res = ast_mutex_trylock(&chan->lock_dont_use);
|
2008-05-13 23:54:01 +00:00
|
|
|
#endif
|
2006-04-15 08:07:50 +00:00
|
|
|
|
|
|
|
if (option_debug > 2) {
|
|
|
|
#ifdef DEBUG_THREADS
|
2006-10-18 02:46:39 +00:00
|
|
|
int count = 0;
|
2008-06-03 18:26:51 +00:00
|
|
|
if ((count = chan->lock_dont_use.track.reentrancy))
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(3, ":::=== Now have %d locks (recursive)\n", count);
|
2006-04-15 08:07:50 +00:00
|
|
|
#endif
|
|
|
|
if (!res)
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(3, "::::==== Channel %s was locked\n", chan->name);
|
2006-04-15 08:07:50 +00:00
|
|
|
if (res == EBUSY) {
|
|
|
|
/* We failed to lock */
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(3, "::::==== Channel %s failed to lock. Not waiting around...\n", chan->name);
|
2006-04-15 08:07:50 +00:00
|
|
|
}
|
|
|
|
if (res == EDEADLK) {
|
|
|
|
/* We had no lock, so okey any way*/
|
2007-06-14 19:39:12 +00:00
|
|
|
ast_debug(3, "::::==== Channel %s was not locked. Lock would cause deadlock.\n", chan->name);
|
2006-04-15 08:07:50 +00:00
|
|
|
}
|
2007-06-14 19:39:12 +00:00
|
|
|
if (res == EINVAL)
|
|
|
|
ast_debug(3, "::::==== Channel %s lock failed. No mutex.\n", chan->name);
|
2006-04-15 08:07:50 +00:00
|
|
|
}
|
|
|
|
return res;
|
|
|
|
}
|
|
|
|
|
|
|
|
#endif
|
2006-04-19 07:23:22 +00:00
|
|
|
|
|
|
|
/*
|
|
|
|
* Wrappers for various ast_say_*() functions that call the full version
|
|
|
|
* of the same functions.
|
|
|
|
* The proper place would be say.c, but that file is optional and one
|
|
|
|
* must be able to build asterisk even without it (using a loadable 'say'
|
|
|
|
* implementation that only supplies the 'full' version of the functions.
|
|
|
|
*/
|
|
|
|
|
|
|
|
int ast_say_number(struct ast_channel *chan, int num,
|
|
|
|
const char *ints, const char *language, const char *options)
|
|
|
|
{
|
2008-01-28 04:43:14 +00:00
|
|
|
return ast_say_number_full(chan, num, ints, language, options, -1, -1);
|
2006-04-19 07:23:22 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
int ast_say_enumeration(struct ast_channel *chan, int num,
|
|
|
|
const char *ints, const char *language, const char *options)
|
|
|
|
{
|
2008-01-28 04:43:14 +00:00
|
|
|
return ast_say_enumeration_full(chan, num, ints, language, options, -1, -1);
|
2006-04-19 07:23:22 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
int ast_say_digits(struct ast_channel *chan, int num,
|
|
|
|
const char *ints, const char *lang)
|
|
|
|
{
|
2008-01-28 04:43:14 +00:00
|
|
|
return ast_say_digits_full(chan, num, ints, lang, -1, -1);
|
2006-04-19 07:23:22 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
int ast_say_digit_str(struct ast_channel *chan, const char *str,
|
|
|
|
const char *ints, const char *lang)
|
|
|
|
{
|
2008-01-28 04:43:14 +00:00
|
|
|
return ast_say_digit_str_full(chan, str, ints, lang, -1, -1);
|
2006-04-19 07:23:22 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
int ast_say_character_str(struct ast_channel *chan, const char *str,
|
|
|
|
const char *ints, const char *lang)
|
|
|
|
{
|
2008-01-28 04:43:14 +00:00
|
|
|
return ast_say_character_str_full(chan, str, ints, lang, -1, -1);
|
2006-04-19 07:23:22 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
int ast_say_phonetic_str(struct ast_channel *chan, const char *str,
|
|
|
|
const char *ints, const char *lang)
|
|
|
|
{
|
2008-01-28 04:43:14 +00:00
|
|
|
return ast_say_phonetic_str_full(chan, str, ints, lang, -1, -1);
|
2006-04-19 07:23:22 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
int ast_say_digits_full(struct ast_channel *chan, int num,
|
|
|
|
const char *ints, const char *lang, int audiofd, int ctrlfd)
|
|
|
|
{
|
2008-01-28 04:43:14 +00:00
|
|
|
char buf[256];
|
2006-04-19 07:23:22 +00:00
|
|
|
|
2008-01-28 04:43:14 +00:00
|
|
|
snprintf(buf, sizeof(buf), "%d", num);
|
|
|
|
|
|
|
|
return ast_say_digit_str_full(chan, buf, ints, lang, audiofd, ctrlfd);
|
2006-04-19 07:23:22 +00:00
|
|
|
}
|