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/*
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* Asterisk - - An open source telephony toolkit .
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*
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* Copyright ( C ) 1999 - 2006 , Digium , Inc .
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*
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* Mark Spencer < markster @ digium . com >
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*
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* See http : //www.asterisk.org for more information about
* the Asterisk project . Please do not directly contact
* any of the maintainers of this project for assistance ;
* the project provides a web site , mailing lists and IRC
* channels for your use .
*
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* This program is free software , distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree .
*/
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/*! \file
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*
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* \ brief Channel Management
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*
* \ author Mark Spencer < markster @ digium . com >
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*/
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# include "asterisk.h"
ASTERISK_FILE_VERSION ( __FILE__ , " $Revision$ " )
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# include "asterisk/_private.h"
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# include <sys/time.h>
# include <signal.h>
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# include <math.h>
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# include "asterisk/paths.h" /* use ast_config_AST_SYSTEM_NAME */
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# include "asterisk/pbx.h"
# include "asterisk/frame.h"
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# include "asterisk/mod_format.h"
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# include "asterisk/sched.h"
# include "asterisk/channel.h"
# include "asterisk/musiconhold.h"
# include "asterisk/say.h"
# include "asterisk/file.h"
# include "asterisk/cli.h"
# include "asterisk/translate.h"
# include "asterisk/manager.h"
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# include "asterisk/cel.h"
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# include "asterisk/chanvars.h"
# include "asterisk/linkedlists.h"
# include "asterisk/indications.h"
# include "asterisk/monitor.h"
# include "asterisk/causes.h"
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# include "asterisk/callerid.h"
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# include "asterisk/utils.h"
# include "asterisk/lock.h"
# include "asterisk/app.h"
# include "asterisk/transcap.h"
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# include "asterisk/devicestate.h"
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: https://origsvn.digium.com/svn/asterisk/trunk@39272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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# include "asterisk/threadstorage.h"
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# include "asterisk/slinfactory.h"
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# include "asterisk/audiohook.h"
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# include "asterisk/framehook.h"
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# include "asterisk/timing.h"
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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# include "asterisk/autochan.h"
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# include "asterisk/stringfields.h"
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# include "asterisk/global_datastores.h"
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# include "asterisk/data.h"
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# ifdef HAVE_EPOLL
# include <sys/epoll.h>
# endif
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# if defined(KEEP_TILL_CHANNEL_PARTY_NUMBER_INFO_NEEDED)
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# if defined(HAVE_PRI)
# include "libpri.h"
# endif /* defined(HAVE_PRI) */
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# endif /* defined(KEEP_TILL_CHANNEL_PARTY_NUMBER_INFO_NEEDED) */
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struct ast_epoll_data {
struct ast_channel * chan ;
int which ;
} ;
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/* uncomment if you have problems with 'monitoring' synchronized files */
#if 0
# define MONITOR_CONSTANT_DELAY
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# define MONITOR_DELAY 150 * 8 /*!< 150 ms of MONITORING DELAY */
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# endif
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/*! \brief Prevent new channel allocation if shutting down. */
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static int shutting_down ;
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static int uniqueint ;
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unsigned long global_fin , global_fout ;
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AST_THREADSTORAGE ( state2str_threadbuf ) ;
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# define STATE2STR_BUFSIZE 32
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/*! Default amount of time to use when emulating a digit as a begin and end
* 100 ms */
# define AST_DEFAULT_EMULATE_DTMF_DURATION 100
/*! Minimum allowed digit length - 80ms */
# define AST_MIN_DTMF_DURATION 80
/*! Minimum amount of time between the end of the last digit and the beginning
* of a new one - 45 ms */
# define AST_MIN_DTMF_GAP 45
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/*! \brief List of channel drivers */
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struct chanlist {
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const struct ast_channel_tech * tech ;
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AST_LIST_ENTRY ( chanlist ) list ;
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} ;
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# ifdef CHANNEL_TRACE
/*! \brief Structure to hold channel context backtrace data */
struct ast_chan_trace_data {
int enabled ;
AST_LIST_HEAD_NOLOCK ( , ast_chan_trace ) trace ;
} ;
/*! \brief Structure to save contexts where an ast_chan has been into */
struct ast_chan_trace {
char context [ AST_MAX_CONTEXT ] ;
char exten [ AST_MAX_EXTENSION ] ;
int priority ;
AST_LIST_ENTRY ( ast_chan_trace ) entry ;
} ;
# endif
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/*! \brief the list of registered channel types */
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
static AST_RWLIST_HEAD_STATIC ( backends , chanlist ) ;
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Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
# ifdef LOW_MEMORY
# define NUM_CHANNEL_BUCKETS 61
# else
# define NUM_CHANNEL_BUCKETS 1567
# endif
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#if 0 /* XXX AstData: ast_callerid no longer exists. (Equivalent code not readily apparent.) */
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# define DATA_EXPORT_CALLERID(MEMBER) \
MEMBER ( ast_callerid , cid_dnid , AST_DATA_STRING ) \
MEMBER ( ast_callerid , cid_num , AST_DATA_STRING ) \
MEMBER ( ast_callerid , cid_name , AST_DATA_STRING ) \
MEMBER ( ast_callerid , cid_ani , AST_DATA_STRING ) \
MEMBER ( ast_callerid , cid_pres , AST_DATA_INTEGER ) \
MEMBER ( ast_callerid , cid_ani2 , AST_DATA_INTEGER ) \
MEMBER ( ast_callerid , cid_tag , AST_DATA_STRING )
AST_DATA_STRUCTURE ( ast_callerid , DATA_EXPORT_CALLERID ) ;
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# endif
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# define DATA_EXPORT_CHANNEL(MEMBER) \
MEMBER ( ast_channel , blockproc , AST_DATA_STRING ) \
MEMBER ( ast_channel , appl , AST_DATA_STRING ) \
MEMBER ( ast_channel , data , AST_DATA_STRING ) \
MEMBER ( ast_channel , name , AST_DATA_STRING ) \
MEMBER ( ast_channel , language , AST_DATA_STRING ) \
MEMBER ( ast_channel , musicclass , AST_DATA_STRING ) \
MEMBER ( ast_channel , accountcode , AST_DATA_STRING ) \
MEMBER ( ast_channel , peeraccount , AST_DATA_STRING ) \
MEMBER ( ast_channel , userfield , AST_DATA_STRING ) \
MEMBER ( ast_channel , call_forward , AST_DATA_STRING ) \
MEMBER ( ast_channel , uniqueid , AST_DATA_STRING ) \
MEMBER ( ast_channel , linkedid , AST_DATA_STRING ) \
MEMBER ( ast_channel , parkinglot , AST_DATA_STRING ) \
MEMBER ( ast_channel , hangupsource , AST_DATA_STRING ) \
MEMBER ( ast_channel , dialcontext , AST_DATA_STRING ) \
MEMBER ( ast_channel , rings , AST_DATA_INTEGER ) \
MEMBER ( ast_channel , priority , AST_DATA_INTEGER ) \
MEMBER ( ast_channel , macropriority , AST_DATA_INTEGER ) \
MEMBER ( ast_channel , adsicpe , AST_DATA_INTEGER ) \
MEMBER ( ast_channel , fin , AST_DATA_UNSIGNED_INTEGER ) \
MEMBER ( ast_channel , fout , AST_DATA_UNSIGNED_INTEGER ) \
MEMBER ( ast_channel , emulate_dtmf_duration , AST_DATA_UNSIGNED_INTEGER ) \
MEMBER ( ast_channel , visible_indication , AST_DATA_INTEGER ) \
MEMBER ( ast_channel , context , AST_DATA_STRING ) \
MEMBER ( ast_channel , exten , AST_DATA_STRING ) \
MEMBER ( ast_channel , macrocontext , AST_DATA_STRING ) \
MEMBER ( ast_channel , macroexten , AST_DATA_STRING )
AST_DATA_STRUCTURE ( ast_channel , DATA_EXPORT_CHANNEL ) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
/*! \brief All active channels on the system */
static struct ao2_container * channels ;
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/*! \brief map AST_CAUSE's to readable string representations
*
* \ ref causes . h
*/
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static const struct {
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int cause ;
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const char * name ;
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const char * desc ;
} causes [ ] = {
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{ AST_CAUSE_UNALLOCATED , " UNALLOCATED " , " Unallocated (unassigned) number " } ,
{ AST_CAUSE_NO_ROUTE_TRANSIT_NET , " NO_ROUTE_TRANSIT_NET " , " No route to specified transmit network " } ,
{ AST_CAUSE_NO_ROUTE_DESTINATION , " NO_ROUTE_DESTINATION " , " No route to destination " } ,
{ AST_CAUSE_CHANNEL_UNACCEPTABLE , " CHANNEL_UNACCEPTABLE " , " Channel unacceptable " } ,
{ AST_CAUSE_CALL_AWARDED_DELIVERED , " CALL_AWARDED_DELIVERED " , " Call awarded and being delivered in an established channel " } ,
{ AST_CAUSE_NORMAL_CLEARING , " NORMAL_CLEARING " , " Normal Clearing " } ,
{ AST_CAUSE_USER_BUSY , " USER_BUSY " , " User busy " } ,
{ AST_CAUSE_NO_USER_RESPONSE , " NO_USER_RESPONSE " , " No user responding " } ,
{ AST_CAUSE_NO_ANSWER , " NO_ANSWER " , " User alerting, no answer " } ,
{ AST_CAUSE_CALL_REJECTED , " CALL_REJECTED " , " Call Rejected " } ,
{ AST_CAUSE_NUMBER_CHANGED , " NUMBER_CHANGED " , " Number changed " } ,
{ AST_CAUSE_DESTINATION_OUT_OF_ORDER , " DESTINATION_OUT_OF_ORDER " , " Destination out of order " } ,
{ AST_CAUSE_INVALID_NUMBER_FORMAT , " INVALID_NUMBER_FORMAT " , " Invalid number format " } ,
{ AST_CAUSE_FACILITY_REJECTED , " FACILITY_REJECTED " , " Facility rejected " } ,
{ AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY , " RESPONSE_TO_STATUS_ENQUIRY " , " Response to STATus ENQuiry " } ,
{ AST_CAUSE_NORMAL_UNSPECIFIED , " NORMAL_UNSPECIFIED " , " Normal, unspecified " } ,
{ AST_CAUSE_NORMAL_CIRCUIT_CONGESTION , " NORMAL_CIRCUIT_CONGESTION " , " Circuit/channel congestion " } ,
{ AST_CAUSE_NETWORK_OUT_OF_ORDER , " NETWORK_OUT_OF_ORDER " , " Network out of order " } ,
{ AST_CAUSE_NORMAL_TEMPORARY_FAILURE , " NORMAL_TEMPORARY_FAILURE " , " Temporary failure " } ,
{ AST_CAUSE_SWITCH_CONGESTION , " SWITCH_CONGESTION " , " Switching equipment congestion " } ,
{ AST_CAUSE_ACCESS_INFO_DISCARDED , " ACCESS_INFO_DISCARDED " , " Access information discarded " } ,
{ AST_CAUSE_REQUESTED_CHAN_UNAVAIL , " REQUESTED_CHAN_UNAVAIL " , " Requested channel not available " } ,
{ AST_CAUSE_PRE_EMPTED , " PRE_EMPTED " , " Pre-empted " } ,
{ AST_CAUSE_FACILITY_NOT_SUBSCRIBED , " FACILITY_NOT_SUBSCRIBED " , " Facility not subscribed " } ,
{ AST_CAUSE_OUTGOING_CALL_BARRED , " OUTGOING_CALL_BARRED " , " Outgoing call barred " } ,
{ AST_CAUSE_INCOMING_CALL_BARRED , " INCOMING_CALL_BARRED " , " Incoming call barred " } ,
{ AST_CAUSE_BEARERCAPABILITY_NOTAUTH , " BEARERCAPABILITY_NOTAUTH " , " Bearer capability not authorized " } ,
{ AST_CAUSE_BEARERCAPABILITY_NOTAVAIL , " BEARERCAPABILITY_NOTAVAIL " , " Bearer capability not available " } ,
{ AST_CAUSE_BEARERCAPABILITY_NOTIMPL , " BEARERCAPABILITY_NOTIMPL " , " Bearer capability not implemented " } ,
{ AST_CAUSE_CHAN_NOT_IMPLEMENTED , " CHAN_NOT_IMPLEMENTED " , " Channel not implemented " } ,
{ AST_CAUSE_FACILITY_NOT_IMPLEMENTED , " FACILITY_NOT_IMPLEMENTED " , " Facility not implemented " } ,
{ AST_CAUSE_INVALID_CALL_REFERENCE , " INVALID_CALL_REFERENCE " , " Invalid call reference value " } ,
{ AST_CAUSE_INCOMPATIBLE_DESTINATION , " INCOMPATIBLE_DESTINATION " , " Incompatible destination " } ,
{ AST_CAUSE_INVALID_MSG_UNSPECIFIED , " INVALID_MSG_UNSPECIFIED " , " Invalid message unspecified " } ,
{ AST_CAUSE_MANDATORY_IE_MISSING , " MANDATORY_IE_MISSING " , " Mandatory information element is missing " } ,
{ AST_CAUSE_MESSAGE_TYPE_NONEXIST , " MESSAGE_TYPE_NONEXIST " , " Message type nonexist. " } ,
{ AST_CAUSE_WRONG_MESSAGE , " WRONG_MESSAGE " , " Wrong message " } ,
{ AST_CAUSE_IE_NONEXIST , " IE_NONEXIST " , " Info. element nonexist or not implemented " } ,
{ AST_CAUSE_INVALID_IE_CONTENTS , " INVALID_IE_CONTENTS " , " Invalid information element contents " } ,
{ AST_CAUSE_WRONG_CALL_STATE , " WRONG_CALL_STATE " , " Message not compatible with call state " } ,
{ AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE , " RECOVERY_ON_TIMER_EXPIRE " , " Recover on timer expiry " } ,
{ AST_CAUSE_MANDATORY_IE_LENGTH_ERROR , " MANDATORY_IE_LENGTH_ERROR " , " Mandatory IE length error " } ,
{ AST_CAUSE_PROTOCOL_ERROR , " PROTOCOL_ERROR " , " Protocol error, unspecified " } ,
{ AST_CAUSE_INTERWORKING , " INTERWORKING " , " Interworking, unspecified " } ,
2005-07-05 17:16:17 +00:00
} ;
2006-02-11 07:06:47 +00:00
struct ast_variable * ast_channeltype_list ( void )
{
struct chanlist * cl ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
struct ast_variable * var = NULL , * prev = NULL ;
AST_RWLIST_RDLOCK ( & backends ) ;
AST_RWLIST_TRAVERSE ( & backends , cl , list ) {
2006-02-11 07:06:47 +00:00
if ( prev ) {
2007-08-29 20:55:40 +00:00
if ( ( prev - > next = ast_variable_new ( cl - > tech - > type , cl - > tech - > description , " " ) ) )
2006-02-11 07:06:47 +00:00
prev = prev - > next ;
} else {
2007-08-29 20:55:40 +00:00
var = ast_variable_new ( cl - > tech - > type , cl - > tech - > description , " " ) ;
2006-02-11 07:06:47 +00:00
prev = var ;
}
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_UNLOCK ( & backends ) ;
2006-02-11 07:06:47 +00:00
return var ;
}
2010-07-08 14:48:42 +00:00
static void channel_data_add_flags ( struct ast_data * tree ,
struct ast_channel * chan )
{
ast_data_add_bool ( tree , " DEFER_DTMF " , ast_test_flag ( chan , AST_FLAG_DEFER_DTMF ) ) ;
ast_data_add_bool ( tree , " WRITE_INT " , ast_test_flag ( chan , AST_FLAG_WRITE_INT ) ) ;
ast_data_add_bool ( tree , " BLOCKING " , ast_test_flag ( chan , AST_FLAG_BLOCKING ) ) ;
ast_data_add_bool ( tree , " ZOMBIE " , ast_test_flag ( chan , AST_FLAG_ZOMBIE ) ) ;
ast_data_add_bool ( tree , " EXCEPTION " , ast_test_flag ( chan , AST_FLAG_EXCEPTION ) ) ;
ast_data_add_bool ( tree , " MOH " , ast_test_flag ( chan , AST_FLAG_MOH ) ) ;
ast_data_add_bool ( tree , " SPYING " , ast_test_flag ( chan , AST_FLAG_SPYING ) ) ;
ast_data_add_bool ( tree , " NBRIDGE " , ast_test_flag ( chan , AST_FLAG_NBRIDGE ) ) ;
ast_data_add_bool ( tree , " IN_AUTOLOOP " , ast_test_flag ( chan , AST_FLAG_IN_AUTOLOOP ) ) ;
ast_data_add_bool ( tree , " OUTGOING " , ast_test_flag ( chan , AST_FLAG_OUTGOING ) ) ;
ast_data_add_bool ( tree , " IN_DTMF " , ast_test_flag ( chan , AST_FLAG_IN_DTMF ) ) ;
ast_data_add_bool ( tree , " EMULATE_DTMF " , ast_test_flag ( chan , AST_FLAG_EMULATE_DTMF ) ) ;
ast_data_add_bool ( tree , " END_DTMF_ONLY " , ast_test_flag ( chan , AST_FLAG_END_DTMF_ONLY ) ) ;
ast_data_add_bool ( tree , " ANSWERED_ELSEWHERE " , ast_test_flag ( chan , AST_FLAG_ANSWERED_ELSEWHERE ) ) ;
ast_data_add_bool ( tree , " MASQ_NOSTREAM " , ast_test_flag ( chan , AST_FLAG_MASQ_NOSTREAM ) ) ;
ast_data_add_bool ( tree , " BRIDGE_HANGUP_RUN " , ast_test_flag ( chan , AST_FLAG_BRIDGE_HANGUP_RUN ) ) ;
ast_data_add_bool ( tree , " BRIDGE_HANGUP_DONT " , ast_test_flag ( chan , AST_FLAG_BRIDGE_HANGUP_DONT ) ) ;
ast_data_add_bool ( tree , " DISABLE_WORKAROUNDS " , ast_test_flag ( chan , AST_FLAG_DISABLE_WORKAROUNDS ) ) ;
}
2010-07-14 15:48:36 +00:00
# if defined(KEEP_TILL_CHANNEL_PARTY_NUMBER_INFO_NEEDED)
2010-07-08 22:05:40 +00:00
static const char * party_number_ton2str ( int ton )
2010-07-08 14:48:42 +00:00
{
2010-07-08 19:12:55 +00:00
# if defined(HAVE_PRI)
2010-07-08 22:05:40 +00:00
switch ( ( ton > > 4 ) & 0x07 ) {
2010-07-08 14:48:42 +00:00
case PRI_TON_INTERNATIONAL :
2010-07-08 22:05:40 +00:00
return " International " ;
2010-07-08 14:48:42 +00:00
case PRI_TON_NATIONAL :
2010-07-08 22:05:40 +00:00
return " National " ;
2010-07-08 14:48:42 +00:00
case PRI_TON_NET_SPECIFIC :
2010-07-08 22:05:40 +00:00
return " Network Specific " ;
2010-07-08 14:48:42 +00:00
case PRI_TON_SUBSCRIBER :
2010-07-08 22:05:40 +00:00
return " Subscriber " ;
2010-07-08 14:48:42 +00:00
case PRI_TON_ABBREVIATED :
2010-07-08 22:05:40 +00:00
return " Abbreviated " ;
2010-07-08 14:48:42 +00:00
case PRI_TON_RESERVED :
2010-07-08 22:05:40 +00:00
return " Reserved " ;
2010-07-08 14:48:42 +00:00
case PRI_TON_UNKNOWN :
default :
2010-07-08 22:05:40 +00:00
break ;
2010-07-08 14:48:42 +00:00
}
2010-07-08 19:12:55 +00:00
# endif /* defined(HAVE_PRI) */
2010-07-08 22:05:40 +00:00
return " Unknown " ;
}
2010-07-14 15:48:36 +00:00
# endif /* defined(KEEP_TILL_CHANNEL_PARTY_NUMBER_INFO_NEEDED) */
2010-07-08 22:05:40 +00:00
2010-07-14 15:48:36 +00:00
# if defined(KEEP_TILL_CHANNEL_PARTY_NUMBER_INFO_NEEDED)
2010-07-08 22:05:40 +00:00
static const char * party_number_plan2str ( int plan )
{
# if defined(HAVE_PRI)
switch ( plan & 0x0F ) {
default :
case PRI_NPI_UNKNOWN :
break ;
case PRI_NPI_E163_E164 :
2010-07-12 17:54:46 +00:00
return " Public (E.163/E.164) " ;
2010-07-08 22:05:40 +00:00
case PRI_NPI_X121 :
2010-07-12 17:54:46 +00:00
return " Data (X.121) " ;
2010-07-08 22:05:40 +00:00
case PRI_NPI_F69 :
2010-07-12 17:54:46 +00:00
return " Telex (F.69) " ;
2010-07-08 22:05:40 +00:00
case PRI_NPI_NATIONAL :
return " National Standard " ;
case PRI_NPI_PRIVATE :
return " Private " ;
case PRI_NPI_RESERVED :
return " Reserved " ;
}
# endif /* defined(HAVE_PRI) */
return " Unknown " ;
2010-07-08 14:48:42 +00:00
}
2010-07-14 15:48:36 +00:00
# endif /* defined(KEEP_TILL_CHANNEL_PARTY_NUMBER_INFO_NEEDED) */
2010-07-08 14:48:42 +00:00
int ast_channel_data_add_structure ( struct ast_data * tree ,
struct ast_channel * chan , int add_bridged )
2010-04-22 18:07:02 +00:00
{
2010-07-08 14:48:42 +00:00
struct ast_channel * bc ;
2010-07-14 15:48:36 +00:00
struct ast_data * data_bridged ;
struct ast_data * data_cdr ;
struct ast_data * data_flags ;
struct ast_data * data_zones ;
struct ast_data * enum_node ;
struct ast_data * data_softhangup ;
#if 0 /* XXX AstData: ast_callerid no longer exists. (Equivalent code not readily apparent.) */
struct ast_data * data_callerid ;
2010-07-08 22:05:40 +00:00
char value_str [ 100 ] ;
2010-07-14 15:48:36 +00:00
# endif
2010-07-08 14:48:42 +00:00
if ( ! tree ) {
return - 1 ;
}
ast_data_add_structure ( ast_channel , tree , chan ) ;
if ( add_bridged ) {
bc = ast_bridged_channel ( chan ) ;
if ( bc ) {
data_bridged = ast_data_add_node ( tree , " bridged " ) ;
if ( ! data_bridged ) {
return - 1 ;
}
ast_channel_data_add_structure ( data_bridged , bc , 0 ) ;
}
}
ast_data_add_codecs ( tree , " oldwriteformat " , chan - > oldwriteformat ) ;
ast_data_add_codecs ( tree , " nativeformats " , chan - > nativeformats ) ;
ast_data_add_codecs ( tree , " readformat " , chan - > readformat ) ;
ast_data_add_codecs ( tree , " writeformat " , chan - > writeformat ) ;
ast_data_add_codecs ( tree , " rawreadformat " , chan - > rawreadformat ) ;
ast_data_add_codecs ( tree , " rawwriteformat " , chan - > rawwriteformat ) ;
/* state */
enum_node = ast_data_add_node ( tree , " state " ) ;
if ( ! enum_node ) {
return - 1 ;
}
ast_data_add_str ( enum_node , " text " , ast_state2str ( chan - > _state ) ) ;
ast_data_add_int ( enum_node , " value " , chan - > _state ) ;
/* hangupcause */
enum_node = ast_data_add_node ( tree , " hangupcause " ) ;
if ( ! enum_node ) {
return - 1 ;
}
ast_data_add_str ( enum_node , " text " , ast_cause2str ( chan - > hangupcause ) ) ;
ast_data_add_int ( enum_node , " value " , chan - > hangupcause ) ;
/* amaflags */
enum_node = ast_data_add_node ( tree , " amaflags " ) ;
if ( ! enum_node ) {
return - 1 ;
}
ast_data_add_str ( enum_node , " text " , ast_cdr_flags2str ( chan - > amaflags ) ) ;
ast_data_add_int ( enum_node , " value " , chan - > amaflags ) ;
/* transfercapability */
enum_node = ast_data_add_node ( tree , " transfercapability " ) ;
if ( ! enum_node ) {
return - 1 ;
}
ast_data_add_str ( enum_node , " text " , ast_transfercapability2str ( chan - > transfercapability ) ) ;
ast_data_add_int ( enum_node , " value " , chan - > transfercapability ) ;
/* _softphangup */
data_softhangup = ast_data_add_node ( tree , " softhangup " ) ;
if ( ! data_softhangup ) {
return - 1 ;
}
ast_data_add_bool ( data_softhangup , " dev " , chan - > _softhangup & AST_SOFTHANGUP_DEV ) ;
ast_data_add_bool ( data_softhangup , " asyncgoto " , chan - > _softhangup & AST_SOFTHANGUP_ASYNCGOTO ) ;
ast_data_add_bool ( data_softhangup , " shutdown " , chan - > _softhangup & AST_SOFTHANGUP_SHUTDOWN ) ;
ast_data_add_bool ( data_softhangup , " timeout " , chan - > _softhangup & AST_SOFTHANGUP_TIMEOUT ) ;
ast_data_add_bool ( data_softhangup , " appunload " , chan - > _softhangup & AST_SOFTHANGUP_APPUNLOAD ) ;
ast_data_add_bool ( data_softhangup , " explicit " , chan - > _softhangup & AST_SOFTHANGUP_EXPLICIT ) ;
ast_data_add_bool ( data_softhangup , " unbridge " , chan - > _softhangup & AST_SOFTHANGUP_UNBRIDGE ) ;
/* channel flags */
data_flags = ast_data_add_node ( tree , " flags " ) ;
if ( ! data_flags ) {
return - 1 ;
}
channel_data_add_flags ( data_flags , chan ) ;
ast_data_add_uint ( tree , " timetohangup " , chan - > whentohangup . tv_sec ) ;
2010-07-14 15:48:36 +00:00
#if 0 /* XXX AstData: ast_callerid no longer exists. (Equivalent code not readily apparent.) */
2010-07-08 14:48:42 +00:00
/* callerid */
data_callerid = ast_data_add_node ( tree , " callerid " ) ;
if ( ! data_callerid ) {
return - 1 ;
}
ast_data_add_structure ( ast_callerid , data_callerid , & ( chan - > cid ) ) ;
/* insert the callerid ton */
enum_node = ast_data_add_node ( data_callerid , " cid_ton " ) ;
if ( ! enum_node ) {
return - 1 ;
}
ast_data_add_int ( enum_node , " value " , chan - > cid . cid_ton ) ;
2010-07-08 22:05:40 +00:00
snprintf ( value_str , sizeof ( value_str ) , " TON: %s/Plan: %s " ,
party_number_ton2str ( chan - > cid . cid_ton ) ,
party_number_plan2str ( chan - > cid . cid_ton ) ) ;
ast_data_add_str ( enum_node , " text " , value_str ) ;
2010-07-14 15:48:36 +00:00
# endif
2010-07-08 14:48:42 +00:00
/* tone zone */
if ( chan - > zone ) {
data_zones = ast_data_add_node ( tree , " zone " ) ;
if ( ! data_zones ) {
return - 1 ;
}
ast_tone_zone_data_add_structure ( data_zones , chan - > zone ) ;
}
/* insert cdr */
data_cdr = ast_data_add_node ( tree , " cdr " ) ;
if ( ! data_cdr ) {
return - 1 ;
}
ast_cdr_data_add_structure ( data_cdr , chan - > cdr , 1 ) ;
return 0 ;
2010-04-22 18:07:02 +00:00
}
int ast_channel_data_cmp_structure ( const struct ast_data_search * tree ,
struct ast_channel * chan , const char * structure_name )
{
return ast_data_search_cmp_structure ( tree , ast_channel , chan , structure_name ) ;
}
2007-02-24 20:29:41 +00:00
/*! \brief Show channel types - CLI command */
2007-10-11 19:03:06 +00:00
static char * handle_cli_core_show_channeltypes ( struct ast_cli_entry * e , int cmd , struct ast_cli_args * a )
2004-12-31 00:04:41 +00:00
{
2006-01-04 09:22:45 +00:00
# define FORMAT "%-10.10s %-40.40s %-12.12s %-12.12s %-12.12s\n"
2005-12-24 05:18:40 +00:00
struct chanlist * cl ;
2006-01-04 09:22:45 +00:00
int count_chan = 0 ;
2007-10-11 19:03:06 +00:00
switch ( cmd ) {
case CLI_INIT :
e - > command = " core show channeltypes " ;
e - > usage =
" Usage: core show channeltypes \n "
" Lists available channel types registered in your \n "
" Asterisk server. \n " ;
return NULL ;
case CLI_GENERATE :
return NULL ;
}
if ( a - > argc ! = 3 )
return CLI_SHOWUSAGE ;
ast_cli ( a - > fd , FORMAT , " Type " , " Description " , " Devicestate " , " Indications " , " Transfer " ) ;
ast_cli ( a - > fd , FORMAT , " ---------- " , " ----------- " , " ----------- " , " ----------- " , " -------- " ) ;
2008-01-24 03:30:56 +00:00
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_RDLOCK ( & backends ) ;
AST_RWLIST_TRAVERSE ( & backends , cl , list ) {
2007-10-11 19:03:06 +00:00
ast_cli ( a - > fd , FORMAT , cl - > tech - > type , cl - > tech - > description ,
2006-04-06 15:40:12 +00:00
( 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
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_UNLOCK ( & backends ) ;
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 ) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_RDLOCK ( & backends ) ;
AST_RWLIST_TRAVERSE ( & backends , cl , list ) {
2007-10-11 19:03:06 +00:00
if ( ! strncasecmp ( a - > word , cl - > tech - > type , wordlen ) & & + + which > a - > n ) {
ret = ast_strdup ( cl - > tech - > type ) ;
break ;
}
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_UNLOCK ( & backends ) ;
2007-10-11 19:03:06 +00:00
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 ;
2009-11-04 14:05:12 +00:00
char buf [ 512 ] ;
2006-01-22 02:23:38 +00:00
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
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_RDLOCK ( & backends ) ;
2006-01-22 02:23:38 +00:00
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_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 ] ) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_UNLOCK ( & backends ) ;
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 "
2009-11-04 14:05:12 +00:00
" Capabilities: %s \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 " ,
2009-11-04 14:05:12 +00:00
ast_getformatname_multiple ( buf , sizeof ( buf ) , ( 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 "
) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_UNLOCK ( & backends ) ;
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
2011-05-25 16:23:11 +00:00
static struct ast_frame * kill_read ( struct ast_channel * chan )
{
/* Hangup channel. */
return NULL ;
}
static struct ast_frame * kill_exception ( struct ast_channel * chan )
{
/* Hangup channel. */
return NULL ;
}
static int kill_write ( struct ast_channel * chan , struct ast_frame * frame )
{
/* Hangup channel. */
return - 1 ;
}
static int kill_fixup ( struct ast_channel * oldchan , struct ast_channel * newchan )
{
/* No problem fixing up the channel. */
return 0 ;
}
static int kill_hangup ( struct ast_channel * chan )
{
chan - > tech_pvt = NULL ;
return 0 ;
}
/*!
* \ brief Kill the channel channel driver technology descriptor .
*
* \ details
* The purpose of this channel technology is to encourage the
* channel to hangup as quickly as possible .
*
* \ note Used by DTMF atxfer and zombie channels .
*/
const struct ast_channel_tech ast_kill_tech = {
. type = " Kill " ,
. description = " Kill channel (should not see this) " ,
. capabilities = - 1 ,
. read = kill_read ,
. exception = kill_exception ,
. write = kill_write ,
. fixup = kill_fixup ,
. hangup = kill_hangup ,
} ;
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 */
2009-05-09 11:30:15 +00:00
static const struct ast_datastore_info ast_chan_trace_datastore_info = {
2008-02-18 04:43:33 +00:00
. 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 ;
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 ;
2010-01-18 22:31:25 +00:00
ast_debug ( 4 , " Hangup time has come: % " PRIi64 " \n " , ast_tvdiff_ms ( chan - > whentohangup , ast_tvnow ( ) ) ) ;
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 ;
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
int ast_check_hangup_locked ( struct ast_channel * chan )
2003-08-11 21:10:52 +00:00
{
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 ;
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
static int ast_channel_softhangup_cb ( void * obj , void * arg , int flags )
{
struct ast_channel * chan = obj ;
ast_softhangup ( chan , AST_SOFTHANGUP_SHUTDOWN ) ;
return 0 ;
}
2002-06-21 01:40:13 +00:00
void ast_begin_shutdown ( int hangup )
{
shutting_down = 1 ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
2002-06-21 01:40:13 +00:00
if ( hangup ) {
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
ao2_callback ( channels , OBJ_NODATA | OBJ_MULTIPLE , ast_channel_softhangup_cb , NULL ) ;
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 )
{
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
return channels ? ao2_container_count ( channels ) : 0 ;
2002-06-21 01:40:13 +00:00
}
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 ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_WRLOCK ( & backends ) ;
2003-03-30 22:55:42 +00:00
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_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 ) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_UNLOCK ( & backends ) ;
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 ) ) ) ) {
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_UNLOCK ( & backends ) ;
1999-12-04 21:35:07 +00:00
return - 1 ;
}
2006-01-11 06:35:34 +00:00
chan - > tech = tech ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_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
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_UNLOCK ( & backends ) ;
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
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_WRLOCK ( & backends ) ;
2005-07-08 21:14:34 +00:00
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_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
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_UNLOCK ( & backends ) ;
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
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_RDLOCK ( & backends ) ;
2005-07-08 21:14:34 +00:00
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_TRAVERSE ( & backends , chanls , list ) {
2006-01-11 05:05:13 +00:00
if ( ! strcasecmp ( name , chanls - > tech - > type ) ) {
ret = chanls - > tech ;
break ;
}
2005-07-08 21:14:34 +00:00
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_UNLOCK ( & backends ) ;
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: https://origsvn.digium.com/svn/asterisk/trunk@39272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 */
2009-11-04 14:05:12 +00:00
format_t ast_best_codec ( format_t fmts )
2001-09-19 14:40:16 +00:00
{
/* This just our opinion, expressed in code. We are asked to choose
the best codec to use , given no information */
int x ;
2009-11-04 14:05:12 +00:00
static const format_t 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 ,
2010-06-16 19:03:24 +00:00
AST_FORMAT_G719 ,
2009-03-24 22:00:58 +00:00
AST_FORMAT_SIREN14 ,
AST_FORMAT_SIREN7 ,
2009-11-04 14:05:12 +00:00
AST_FORMAT_TESTLAW ,
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 */
2010-06-17 17:23:43 +00:00
AST_FORMAT_SPEEX16 ,
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 ,
} ;
2009-11-04 14:05:12 +00:00
char buf [ 512 ] ;
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
}
2009-11-04 14:05:12 +00:00
ast_log ( LOG_WARNING , " Don't know any of %s formats \n " , ast_getformatname_multiple ( buf , sizeof ( buf ) , 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
} ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
static void ast_channel_destructor ( void * obj ) ;
2009-06-26 15:28:53 +00:00
static void ast_dummy_channel_destructor ( void * obj ) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
2005-12-31 16:35:48 +00:00
/*! \brief Create a new channel structure */
2009-06-26 15:28:53 +00:00
static struct ast_channel * attribute_malloc __attribute__ ( ( format ( printf , 13 , 0 ) ) )
2009-05-05 10:34:19 +00:00
__ast_channel_alloc_ap ( int needqueue , int state , const char * cid_num , const char * cid_name ,
const char * acctcode , const char * exten , const char * context ,
2009-06-26 15:28:53 +00:00
const char * linkedid , const int amaflag , const char * file , int line ,
const char * function , const char * name_fmt , va_list ap1 , va_list ap2 )
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 ;
2010-09-28 18:18:02 +00:00
char * tech = " " , * tech2 = NULL ;
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
2009-05-05 10:34:19 +00:00
# if defined(REF_DEBUG)
2010-05-21 22:46:52 +00:00
tmp = __ao2_alloc_debug ( sizeof ( * tmp ) , ast_channel_destructor , " " , file , line ,
function , 1 ) ;
2009-05-05 10:34:19 +00:00
# elif defined(__AST_DEBUG_MALLOC)
2010-05-21 22:46:52 +00:00
tmp = __ao2_alloc_debug ( sizeof ( * tmp ) , ast_channel_destructor , " " , file , line ,
function , 0 ) ;
2009-05-05 10:34:19 +00:00
# else
2010-05-21 22:46:52 +00:00
tmp = ao2_alloc ( sizeof ( * tmp ) , ast_channel_destructor ) ;
# endif
if ( ! tmp ) {
/* Channel structure allocation failure. */
2005-03-04 06:47:24 +00:00
return NULL ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
}
2010-05-21 22:46:52 +00:00
/*
* Init file descriptors to unopened state so
* the destructor can know not to close them .
*/
tmp - > timingfd = - 1 ;
for ( x = 0 ; x < ARRAY_LEN ( tmp - > alertpipe ) ; + + x ) {
tmp - > alertpipe [ x ] = - 1 ;
}
for ( x = 0 ; x < ARRAY_LEN ( tmp - > fds ) ; + + x ) {
tmp - > fds [ x ] = - 1 ;
}
# ifdef HAVE_EPOLL
tmp - > epfd = epoll_create ( 25 ) ;
2009-05-05 10:34:19 +00:00
# endif
2005-03-04 06:47:24 +00:00
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 " ) ;
2009-06-11 15:40:03 +00:00
return ast_channel_unref ( tmp ) ;
2005-03-04 06:47:24 +00:00
}
2010-07-14 15:48:36 +00:00
ast_party_dialed_init ( & tmp - > dialed ) ;
ast_party_caller_init ( & tmp - > caller ) ;
ast_party_connected_line_init ( & tmp - > connected ) ;
ast_party_redirecting_init ( & tmp - > redirecting ) ;
2009-04-10 16:06:22 +00:00
if ( cid_name ) {
2010-07-14 15:48:36 +00:00
tmp - > caller . id . name . valid = 1 ;
tmp - > caller . id . name . str = ast_strdup ( cid_name ) ;
if ( ! tmp - > caller . id . name . str ) {
2009-06-10 18:58:12 +00:00
return ast_channel_unref ( tmp ) ;
2009-04-10 16:06:22 +00:00
}
}
if ( cid_num ) {
2010-07-14 15:48:36 +00:00
tmp - > caller . id . number . valid = 1 ;
tmp - > caller . id . number . str = ast_strdup ( cid_num ) ;
if ( ! tmp - > caller . id . number . str ) {
2009-06-10 18:58:12 +00:00
return ast_channel_unref ( tmp ) ;
2009-04-10 16:06:22 +00:00
}
2009-04-03 22:41:46 +00:00
}
2009-03-27 19:10:32 +00:00
if ( ( tmp - > timer = ast_timer_open ( ) ) ) {
2010-11-08 21:59:45 +00:00
if ( strcmp ( ast_timer_get_name ( tmp - > timer ) , " timerfd " ) ) {
needqueue = 0 ;
}
2009-03-27 19:10:32 +00:00
tmp - > timingfd = ast_timer_fd ( tmp - > timer ) ;
2005-03-04 06:47:24 +00:00
}
if ( needqueue ) {
if ( pipe ( tmp - > alertpipe ) ) {
2009-11-09 15:38:38 +00:00
ast_log ( LOG_WARNING , " Channel allocation failed: Can't create alert pipe! Try increasing max file descriptors with ulimit -n \n " ) ;
2010-05-21 22:46:52 +00:00
return ast_channel_unref ( tmp ) ;
2005-03-04 06:47:24 +00:00
} 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 ) ) ;
2010-05-21 22:46:52 +00:00
return ast_channel_unref ( tmp ) ;
2008-06-12 15:14:37 +00:00
}
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 ) ) ;
2010-05-21 22:46:52 +00:00
return ast_channel_unref ( tmp ) ;
2008-06-12 15:14:37 +00:00
}
1999-12-04 21:35:07 +00:00
}
2010-05-21 22:46:52 +00:00
}
/*
* This is the last place the channel constructor can fail .
*
* The destructor takes advantage of this fact to ensure that the
* AST_CEL_CHANNEL_END is not posted if we have not posted the
* AST_CEL_CHANNEL_START yet .
*/
if ( ( ast_string_field_init ( tmp , 128 ) ) ) {
return ast_channel_unref ( tmp ) ;
}
2005-03-04 06:47:24 +00:00
/* 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 ) ;
2006-11-07 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 ) ,
2010-05-21 22:46:52 +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 ,
2010-05-21 22:46:52 +00:00
( long ) time ( NULL ) , ast_atomic_fetchadd_int ( & uniqueint , 1 ) ) ;
2006-08-21 19:35:28 +00:00
}
2009-06-26 15:28:53 +00:00
if ( ! ast_strlen_zero ( linkedid ) ) {
ast_string_field_set ( tmp , linkedid , linkedid ) ;
2010-05-21 22:46:52 +00:00
} else {
2009-06-26 15:28:53 +00:00
ast_string_field_set ( tmp , linkedid , tmp - > uniqueid ) ;
}
2006-11-07 21:47:49 +00:00
if ( ! ast_strlen_zero ( name_fmt ) ) {
2010-09-28 18:18:02 +00:00
char * slash , * slash2 ;
2006-11-07 21:47:49 +00:00
/* 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 .
*/
ast_string_field_build_va ( tmp , name , name_fmt , ap1 , ap2 ) ;
2010-04-26 21:13:35 +00:00
tech = ast_strdupa ( tmp - > name ) ;
if ( ( slash = strchr ( tech , ' / ' ) ) ) {
2010-09-28 18:18:02 +00:00
if ( ( slash2 = strchr ( slash + 1 , ' / ' ) ) ) {
tech2 = slash + 1 ;
* slash2 = ' \0 ' ;
}
2010-04-26 21:13:35 +00:00
* slash = ' \0 ' ;
}
2010-05-21 22:46:52 +00:00
} else {
/*
* Start the string with ' - ' so it becomes an empty string
* in the destructor .
*/
ast_string_field_set ( tmp , name , " -**Unknown** " ) ;
2006-11-07 21:47:49 +00:00
}
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 ;
2009-06-26 15:28:53 +00:00
2007-03-30 14:37:21 +00:00
tmp - > cdr = ast_cdr_alloc ( ) ;
ast_cdr_init ( tmp - > cdr , tmp ) ;
ast_cdr_start ( tmp - > cdr ) ;
2009-06-26 15:28:53 +00:00
ast_cel_report_event ( tmp , AST_CEL_CHANNEL_START , NULL , NULL , NULL ) ;
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
2006-04-10 23:29:50 +00:00
AST_LIST_HEAD_INIT_NOLOCK ( & tmp - > datastores ) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_LIST_HEAD_INIT_NOLOCK ( & tmp - > autochans ) ;
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 ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
ao2_link ( channels , tmp ) ;
2006-08-21 19:35:28 +00:00
2010-05-21 22:46:52 +00:00
/*
* And now , since the channel structure is built , and has its name , let ' s
2007-12-24 19:43:36 +00:00
* 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 !
*/
2010-09-28 18:18:02 +00:00
if ( ast_get_channel_tech ( tech ) | | ( tech2 & & ast_get_channel_tech ( tech2 ) ) ) {
2009-11-13 20:42:03 +00:00
ast_manager_event ( tmp , EVENT_FLAG_CALL , " Newchannel " ,
2007-12-24 19:43:36 +00:00
" Channel: %s \r \n "
" ChannelState: %d \r \n "
" ChannelStateDesc: %s \r \n "
" CallerIDNum: %s \r \n "
" CallerIDName: %s \r \n "
" AccountCode: %s \r \n "
2009-01-26 12:32:30 +00:00
" Exten: %s \r \n "
" Context: %s \r \n "
2007-12-24 19:43:36 +00:00
" Uniqueid: %s \r \n " ,
tmp - > name ,
state ,
ast_state2str ( state ) ,
S_OR ( cid_num , " " ) ,
S_OR ( cid_name , " " ) ,
tmp - > accountcode ,
2009-01-26 12:32:30 +00:00
S_OR ( exten , " " ) ,
S_OR ( context , " " ) ,
2007-12-24 19:43:36 +00:00
tmp - > uniqueid ) ;
}
1999-12-04 21:35:07 +00:00
return tmp ;
}
2009-05-05 10:34:19 +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 ,
2009-06-26 15:28:53 +00:00
const char * linkedid , const int amaflag ,
const char * file , int line , const char * function ,
const char * name_fmt , . . . )
2009-05-05 10:34:19 +00:00
{
va_list ap1 , ap2 ;
struct ast_channel * result ;
va_start ( ap1 , name_fmt ) ;
va_start ( ap2 , name_fmt ) ;
result = __ast_channel_alloc_ap ( needqueue , state , cid_num , cid_name , acctcode , exten , context ,
2009-06-26 15:28:53 +00:00
linkedid , amaflag , file , line , function , name_fmt , ap1 , ap2 ) ;
2009-05-05 10:34:19 +00:00
va_end ( ap1 ) ;
va_end ( ap2 ) ;
return result ;
}
2009-06-26 15:28:53 +00:00
/* only do the minimum amount of work needed here to make a channel
* structure that can be used to expand channel vars */
2009-06-29 17:56:29 +00:00
# if defined(REF_DEBUG) || defined(__AST_DEBUG_MALLOC)
struct ast_channel * __ast_dummy_channel_alloc ( const char * file , int line , const char * function )
# else
2009-06-26 15:28:53 +00:00
struct ast_channel * ast_dummy_channel_alloc ( void )
2009-06-29 17:56:29 +00:00
# endif
2009-06-26 15:28:53 +00:00
{
struct ast_channel * tmp ;
struct varshead * headp ;
# if defined(REF_DEBUG)
2010-05-21 22:46:52 +00:00
tmp = __ao2_alloc_debug ( sizeof ( * tmp ) , ast_dummy_channel_destructor , " dummy channel " ,
file , line , function , 1 ) ;
2009-06-26 15:28:53 +00:00
# elif defined(__AST_DEBUG_MALLOC)
2010-05-21 22:46:52 +00:00
tmp = __ao2_alloc_debug ( sizeof ( * tmp ) , ast_dummy_channel_destructor , " dummy channel " ,
file , line , function , 0 ) ;
2009-06-26 15:28:53 +00:00
# else
2010-05-21 22:46:52 +00:00
tmp = ao2_alloc ( sizeof ( * tmp ) , ast_dummy_channel_destructor ) ;
# endif
if ( ! tmp ) {
/* Dummy channel structure allocation failure. */
2009-06-26 15:28:53 +00:00
return NULL ;
}
if ( ( ast_string_field_init ( tmp , 128 ) ) ) {
2010-05-21 22:46:52 +00:00
return ast_channel_unref ( tmp ) ;
2009-06-26 15:28:53 +00:00
}
headp = & tmp - > varshead ;
AST_LIST_HEAD_INIT_NOLOCK ( headp ) ;
return tmp ;
}
2009-06-16 18:54:30 +00:00
static int __ast_queue_frame ( struct ast_channel * chan , struct ast_frame * fin , int head , struct ast_frame * after )
2002-06-21 01:40:13 +00:00
{
struct ast_frame * f ;
2006-08-29 20:50:36 +00:00
struct ast_frame * cur ;
2009-06-16 18:54:30 +00:00
unsigned int new_frames = 0 ;
unsigned int new_voice_frames = 0 ;
unsigned int queued_frames = 0 ;
unsigned int queued_voice_frames = 0 ;
AST_LIST_HEAD_NOLOCK ( , ast_frame ) frames ;
2008-12-12 13:55:30 +00:00
2006-04-26 21:44:25 +00:00
ast_channel_lock ( chan ) ;
2006-08-29 20:50:36 +00:00
2010-11-22 19:36:10 +00:00
/*
* Check the last frame on the queue if we are queuing the new
* frames after it .
*/
cur = AST_LIST_LAST ( & chan - > readq ) ;
if ( cur & & cur - > frametype = = AST_FRAME_CONTROL & & ! head & & ( ! after | | after = = cur ) ) {
switch ( cur - > subclass . integer ) {
case AST_CONTROL_END_OF_Q :
if ( fin - > frametype = = AST_FRAME_CONTROL
& & fin - > subclass . integer = = AST_CONTROL_HANGUP ) {
/*
* Destroy the end - of - Q marker frame so we can queue the hangup
* frame in its place .
*/
AST_LIST_REMOVE ( & chan - > readq , cur , frame_list ) ;
ast_frfree ( cur ) ;
/*
* This has degenerated to a normal queue append anyway . Since
* we just destroyed the last frame in the queue we must make
* sure that " after " is NULL or bad things will happen .
*/
after = NULL ;
break ;
}
/* Fall through */
case AST_CONTROL_HANGUP :
/* Don't queue anything. */
ast_channel_unlock ( chan ) ;
return 0 ;
default :
break ;
}
2006-08-29 20:50:36 +00:00
}
2010-11-22 19:36:10 +00:00
/* Build copies of all the new frames and count them */
2009-06-16 18:54:30 +00:00
AST_LIST_HEAD_INIT_NOLOCK ( & frames ) ;
for ( cur = fin ; cur ; cur = AST_LIST_NEXT ( cur , frame_list ) ) {
if ( ! ( f = ast_frdup ( cur ) ) ) {
2010-09-21 19:48:53 +00:00
if ( AST_LIST_FIRST ( & frames ) ) {
ast_frfree ( AST_LIST_FIRST ( & frames ) ) ;
}
2009-11-25 22:33:02 +00:00
ast_channel_unlock ( chan ) ;
2009-06-16 18:54:30 +00:00
return - 1 ;
}
AST_LIST_INSERT_TAIL ( & frames , f , frame_list ) ;
new_frames + + ;
if ( f - > frametype = = AST_FRAME_VOICE ) {
new_voice_frames + + ;
}
}
2006-08-29 20:50:36 +00:00
/* Count how many frames exist on the queue */
AST_LIST_TRAVERSE ( & chan - > readq , cur , frame_list ) {
2009-06-16 18:54:30 +00:00
queued_frames + + ;
if ( cur - > frametype = = AST_FRAME_VOICE ) {
queued_voice_frames + + ;
}
2002-06-21 01:40:13 +00:00
}
2006-08-29 20:50:36 +00:00
2010-02-01 17:53:39 +00:00
if ( ( queued_frames + new_frames > 128 | | queued_voice_frames + new_voice_frames > 96 ) ) {
int count = 0 ;
ast_log ( LOG_WARNING , " Exceptionally long %squeue length queuing to %s \n " , queued_frames + new_frames > 128 ? " " : " voice " , chan - > name ) ;
AST_LIST_TRAVERSE_SAFE_BEGIN ( & chan - > readq , cur , frame_list ) {
/* Save the most recent frame */
if ( ! AST_LIST_NEXT ( cur , frame_list ) ) {
break ;
} else if ( cur - > frametype = = AST_FRAME_VOICE | | cur - > frametype = = AST_FRAME_VIDEO | | cur - > frametype = = AST_FRAME_NULL ) {
if ( + + count > 64 ) {
break ;
}
AST_LIST_REMOVE_CURRENT ( frame_list ) ;
ast_frfree ( cur ) ;
}
2003-02-05 21:18:09 +00:00
}
2010-02-01 17:53:39 +00:00
AST_LIST_TRAVERSE_SAFE_END ;
2003-02-05 21:18:09 +00:00
}
2008-12-12 13:55:30 +00:00
2009-06-16 18:54:30 +00:00
if ( after ) {
AST_LIST_INSERT_LIST_AFTER ( & chan - > readq , & frames , after , frame_list ) ;
2008-12-12 13:55:30 +00:00
} else {
2009-06-16 18:54:30 +00:00
if ( head ) {
AST_LIST_APPEND_LIST ( & frames , & chan - > readq , frame_list ) ;
AST_LIST_HEAD_INIT_NOLOCK ( & chan - > readq ) ;
}
AST_LIST_APPEND_LIST ( & chan - > readq , & frames , frame_list ) ;
2008-12-12 13:55:30 +00:00
}
2005-03-04 06:47:24 +00:00
if ( chan - > alertpipe [ 1 ] > - 1 ) {
2011-11-14 19:05:09 +00:00
int blah [ new_frames ] ;
memset ( blah , 1 , sizeof ( blah ) ) ;
if ( write ( chan - > alertpipe [ 1 ] , & blah , sizeof ( blah ) ) ! = ( sizeof ( blah ) ) ) {
2009-06-16 18:54:30 +00:00
ast_log ( LOG_WARNING , " Unable to write to alert pipe on %s (qlen = %d): %s! \n " ,
chan - > name , queued_frames , strerror ( errno ) ) ;
2008-12-12 13:55:30 +00:00
}
2004-03-19 04:23:20 +00:00
} else if ( chan - > timingfd > - 1 ) {
2009-03-27 19:10:32 +00:00
ast_timer_enable_continuous ( chan - > timer ) ;
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
}
2008-12-12 13:55:30 +00:00
2006-04-26 21:44:25 +00:00
ast_channel_unlock ( chan ) ;
2008-12-12 13:55:30 +00:00
2002-06-21 01:40:13 +00:00
return 0 ;
}
2009-06-16 18:54:30 +00:00
int ast_queue_frame ( struct ast_channel * chan , struct ast_frame * fin )
2008-12-12 13:55:30 +00:00
{
2009-06-16 18:54:30 +00:00
return __ast_queue_frame ( chan , fin , 0 , NULL ) ;
2008-12-12 13:55:30 +00:00
}
2009-06-16 18:54:30 +00:00
int ast_queue_frame_head ( struct ast_channel * chan , struct ast_frame * fin )
2008-12-12 13:55:30 +00:00
{
2009-06-16 18:54:30 +00:00
return __ast_queue_frame ( chan , fin , 1 , NULL ) ;
2008-12-12 13:55:30 +00:00
}
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 )
{
2009-11-04 14:05:12 +00:00
struct ast_frame f = { AST_FRAME_CONTROL , . subclass . integer = AST_CONTROL_HANGUP } ;
2008-05-22 16:29:54 +00:00
/* 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
{
2009-11-04 14:05:12 +00:00
struct ast_frame f = { AST_FRAME_CONTROL , . subclass . integer = 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
{
2009-11-04 14:05:12 +00:00
struct ast_frame f = { AST_FRAME_CONTROL , . subclass . integer = control } ;
2006-05-09 14:25:57 +00:00
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 )
{
2009-11-04 14:05:12 +00:00
struct ast_frame f = { AST_FRAME_CONTROL , . subclass . integer = control , . data . ptr = ( void * ) data , . 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
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
struct ast_channel * ast_channel_callback ( ao2_callback_data_fn * cb_fn , void * arg ,
void * data , int ao2_flags )
1999-12-18 07:01:48 +00:00
{
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
return ao2_callback_data ( channels , ao2_flags , cb_fn , arg , data ) ;
}
struct ast_channel_iterator {
2009-10-21 21:08:47 +00:00
/* storage for non-dynamically allocated iterator */
struct ao2_iterator simple_iterator ;
/* pointer to the actual iterator (simple_iterator or a dynamically
* allocated iterator )
*/
struct ao2_iterator * active_iterator ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
} ;
struct ast_channel_iterator * ast_channel_iterator_destroy ( struct ast_channel_iterator * i )
{
2009-10-21 21:08:47 +00:00
ao2_iterator_destroy ( i - > active_iterator ) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
ast_free ( i ) ;
2005-06-06 02:29:18 +00:00
return NULL ;
}
2009-10-21 21:08:47 +00:00
static struct ast_channel_iterator * channel_iterator_search ( const char * name ,
size_t name_len , const char * exten ,
const char * context )
2005-06-06 02:29:18 +00:00
{
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
struct ast_channel_iterator * i ;
2009-10-21 21:08:47 +00:00
struct ast_channel tmp_chan = {
. name = name ,
/* This is sort of a hack. Basically, we're using an arbitrary field
* in ast_channel to pass the name_len for a prefix match . If this
* gets changed , then the compare callback must be changed , too . */
. rings = name_len ,
} ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
if ( ! ( i = ast_calloc ( 1 , sizeof ( * i ) ) ) ) {
return NULL ;
}
2009-10-21 21:08:47 +00:00
if ( exten ) {
ast_copy_string ( tmp_chan . exten , exten , sizeof ( tmp_chan . exten ) ) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
}
2009-10-21 21:08:47 +00:00
if ( context ) {
ast_copy_string ( tmp_chan . context , context , sizeof ( tmp_chan . context ) ) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
}
2009-10-21 21:08:47 +00:00
if ( ! ( i - > active_iterator = ao2_find ( channels , & tmp_chan ,
OBJ_MULTIPLE | ( ( ! ast_strlen_zero ( name ) & & ( name_len = = 0 ) ) ? OBJ_POINTER : 0 ) ) ) ) {
ast_free ( i ) ;
return NULL ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
}
return i ;
2005-07-08 21:14:34 +00:00
}
2009-10-21 21:08:47 +00:00
struct ast_channel_iterator * ast_channel_iterator_by_exten_new ( const char * exten , const char * context )
2005-07-08 21:14:34 +00:00
{
2009-10-21 21:08:47 +00:00
return channel_iterator_search ( NULL , 0 , exten , context ) ;
2005-10-16 16:11:52 +00:00
}
2009-10-21 21:08:47 +00:00
struct ast_channel_iterator * ast_channel_iterator_by_name_new ( const char * name , size_t name_len )
2005-10-16 16:11:52 +00:00
{
2009-10-21 21:08:47 +00:00
return channel_iterator_search ( name , name_len , NULL , NULL ) ;
2005-09-15 16:34:39 +00:00
}
2009-10-21 21:08:47 +00:00
struct ast_channel_iterator * ast_channel_iterator_all_new ( void )
2005-09-15 16:34:39 +00:00
{
2009-10-21 21:08:47 +00:00
struct ast_channel_iterator * i ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
2009-10-21 21:08:47 +00:00
if ( ! ( i = ast_calloc ( 1 , sizeof ( * i ) ) ) ) {
return NULL ;
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
2009-10-21 21:08:47 +00:00
i - > simple_iterator = ao2_iterator_init ( channels , 0 ) ;
i - > active_iterator = & i - > simple_iterator ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
2009-10-21 21:08:47 +00:00
return i ;
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
2009-10-21 21:08:47 +00:00
struct ast_channel * ast_channel_iterator_next ( struct ast_channel_iterator * i )
{
return ao2_iterator_next ( i - > active_iterator ) ;
2004-05-22 04:11:22 +00:00
}
2009-09-25 14:38:41 +00:00
static int ast_channel_cmp_cb ( void * obj , void * arg , int flags )
{
struct ast_channel * chan = obj , * cmp_args = arg ;
size_t name_len ;
int ret = CMP_MATCH ;
/* This is sort of a hack. Basically, we're using an arbitrary field
* in ast_channel to pass the name_len for a prefix match . If this
* gets changed , then the uses of ao2_find ( ) must be changed , too . */
name_len = cmp_args - > rings ;
ast_channel_lock ( chan ) ;
2009-12-15 23:41:20 +00:00
if ( ! ast_strlen_zero ( cmp_args - > name ) ) { /* match by name */
2009-09-25 14:38:41 +00:00
if ( ( ! name_len & & strcasecmp ( chan - > name , cmp_args - > name ) ) | |
( name_len & & strncasecmp ( chan - > name , cmp_args - > name , name_len ) ) ) {
ret = 0 ; /* name match failed */
}
2009-12-15 23:41:20 +00:00
} else if ( ! ast_strlen_zero ( cmp_args - > exten ) ) {
2009-09-25 14:38:41 +00:00
if ( cmp_args - > context & & strcasecmp ( chan - > context , cmp_args - > context ) & &
strcasecmp ( chan - > macrocontext , cmp_args - > context ) ) {
ret = 0 ; /* context match failed */
}
if ( ret & & strcasecmp ( chan - > exten , cmp_args - > exten ) & &
strcasecmp ( chan - > macroexten , cmp_args - > exten ) ) {
ret = 0 ; /* exten match failed */
}
2009-12-15 23:41:20 +00:00
} else if ( ! ast_strlen_zero ( cmp_args - > uniqueid ) ) {
2009-09-25 14:38:41 +00:00
if ( ( ! name_len & & strcasecmp ( chan - > uniqueid , cmp_args - > uniqueid ) ) | |
( name_len & & strncasecmp ( chan - > uniqueid , cmp_args - > uniqueid , name_len ) ) ) {
ret = 0 ; /* uniqueid match failed */
}
2009-12-15 23:41:20 +00:00
} else {
ret = 0 ;
2009-09-25 14:38:41 +00:00
}
ast_channel_unlock ( chan ) ;
return ret ;
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
static struct ast_channel * ast_channel_get_full ( const char * name , size_t name_len ,
2009-10-21 21:08:47 +00:00
const char * exten , const char * context )
2006-07-28 23:36:06 +00:00
{
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
struct ast_channel tmp_chan = {
. name = name ,
/* This is sort of a hack. Basically, we're using an arbitrary field
* in ast_channel to pass the name_len for a prefix match . If this
* gets changed , then the compare callback must be changed , too . */
. rings = name_len ,
} ;
2009-08-10 15:46:39 +00:00
struct ast_channel * chan ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
if ( exten ) {
ast_copy_string ( tmp_chan . exten , exten , sizeof ( tmp_chan . exten ) ) ;
}
if ( context ) {
ast_copy_string ( tmp_chan . context , context , sizeof ( tmp_chan . context ) ) ;
}
2009-09-25 14:38:41 +00:00
if ( ( chan = ao2_find ( channels , & tmp_chan ,
2009-09-25 14:44:40 +00:00
( ! ast_strlen_zero ( name ) & & ( name_len = = 0 ) ) ? OBJ_POINTER : 0 ) ) ) {
2009-08-10 15:46:39 +00:00
return chan ;
}
if ( ! name ) {
return NULL ;
}
/* If name was specified, but the result was NULL,
* try a search on uniqueid , instead . */
{
struct ast_channel tmp_chan2 = {
. uniqueid = name ,
. rings = name_len ,
} ;
return ao2_find ( channels , & tmp_chan2 , 0 ) ;
}
2006-07-28 23:36:06 +00:00
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
struct ast_channel * ast_channel_get_by_name ( const char * name )
2008-11-04 23:23:39 +00:00
{
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
return ast_channel_get_full ( name , 0 , NULL , NULL ) ;
}
2008-11-04 23:23:39 +00:00
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
struct ast_channel * ast_channel_get_by_name_prefix ( const char * name , size_t name_len )
{
return ast_channel_get_full ( name , name_len , NULL , NULL ) ;
}
2008-11-04 23:23:39 +00:00
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
struct ast_channel * ast_channel_get_by_exten ( const char * exten , const char * context )
{
return ast_channel_get_full ( NULL , 0 , exten , context ) ;
2008-11-04 23:23:39 +00:00
}
Merged revisions 264996 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines
Allow ast_safe_sleep to defer specific frames until after the sleep has concluded.
From reviewboard
Background:
A Digium customer discovered a somewhat odd bug. The setup is that parties A
and B are bridged, and party A places party B on hold. While party B is
listening to hold music, he mashes a bunch of DTMF. Party A takes party
B off hold while this is happening, but party B continues to hear hold
music. I could reproduce this about 1 in 5 times.
The issue:
When DTMF features are enabled and a user presses keys, the channel that
the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the
duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read
from the channel during the sleep, the frame is dropped. Thus the
unhold indication is never made to the channel that was originally placed
on hold.
The fix:
Originally, I discussed with Kevin possible ways of fixing the specific
problem reported. However, we determined that the same type of problem
could happen in other situations where ast_safe_sleep() is used. Using
autoservice as a model, I modified ast_safe_sleep_conditional() to
defer specific frame types so they can be re-queued once the sleep has
finished. I made a common function for determining if a frame should
be deferred so that there are not two identical switch blocks to
maintain.
Review: https://reviewboard.asterisk.org/r/674/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-21 16:44:27 +00:00
int ast_is_deferrable_frame ( const struct ast_frame * frame )
{
/* Do not add a default entry in this switch statement. Each new
* frame type should be addressed directly as to whether it should
* be queued up or not .
*/
switch ( frame - > frametype ) {
case AST_FRAME_CONTROL :
case AST_FRAME_TEXT :
case AST_FRAME_IMAGE :
case AST_FRAME_HTML :
return 1 ;
2011-01-12 18:12:08 +00:00
case AST_FRAME_DTMF_END :
Merged revisions 264996 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines
Allow ast_safe_sleep to defer specific frames until after the sleep has concluded.
From reviewboard
Background:
A Digium customer discovered a somewhat odd bug. The setup is that parties A
and B are bridged, and party A places party B on hold. While party B is
listening to hold music, he mashes a bunch of DTMF. Party A takes party
B off hold while this is happening, but party B continues to hear hold
music. I could reproduce this about 1 in 5 times.
The issue:
When DTMF features are enabled and a user presses keys, the channel that
the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the
duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read
from the channel during the sleep, the frame is dropped. Thus the
unhold indication is never made to the channel that was originally placed
on hold.
The fix:
Originally, I discussed with Kevin possible ways of fixing the specific
problem reported. However, we determined that the same type of problem
could happen in other situations where ast_safe_sleep() is used. Using
autoservice as a model, I modified ast_safe_sleep_conditional() to
defer specific frame types so they can be re-queued once the sleep has
finished. I made a common function for determining if a frame should
be deferred so that there are not two identical switch blocks to
maintain.
Review: https://reviewboard.asterisk.org/r/674/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-21 16:44:27 +00:00
case AST_FRAME_DTMF_BEGIN :
case AST_FRAME_VOICE :
case AST_FRAME_VIDEO :
case AST_FRAME_NULL :
case AST_FRAME_IAX :
case AST_FRAME_CNG :
case AST_FRAME_MODEM :
return 0 ;
}
return 0 ;
}
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 ;
2010-01-13 16:31:14 +00:00
struct ast_silence_generator * silgen = NULL ;
int res = 0 ;
Merged revisions 264996 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines
Allow ast_safe_sleep to defer specific frames until after the sleep has concluded.
From reviewboard
Background:
A Digium customer discovered a somewhat odd bug. The setup is that parties A
and B are bridged, and party A places party B on hold. While party B is
listening to hold music, he mashes a bunch of DTMF. Party A takes party
B off hold while this is happening, but party B continues to hear hold
music. I could reproduce this about 1 in 5 times.
The issue:
When DTMF features are enabled and a user presses keys, the channel that
the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the
duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read
from the channel during the sleep, the frame is dropped. Thus the
unhold indication is never made to the channel that was originally placed
on hold.
The fix:
Originally, I discussed with Kevin possible ways of fixing the specific
problem reported. However, we determined that the same type of problem
could happen in other situations where ast_safe_sleep() is used. Using
autoservice as a model, I modified ast_safe_sleep_conditional() to
defer specific frame types so they can be re-queued once the sleep has
finished. I made a common function for determining if a frame should
be deferred so that there are not two identical switch blocks to
maintain.
Review: https://reviewboard.asterisk.org/r/674/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-21 16:44:27 +00:00
AST_LIST_HEAD_NOLOCK ( , ast_frame ) deferred_frames ;
AST_LIST_HEAD_INIT_NOLOCK ( & deferred_frames ) ;
2010-01-13 16:31:14 +00:00
/* If no other generator is present, start silencegen while waiting */
if ( ast_opt_transmit_silence & & ! chan - > generatordata ) {
silgen = ast_channel_start_silence_generator ( chan ) ;
}
2003-02-05 21:18:09 +00:00
2006-01-01 08:45:29 +00:00
while ( ms > 0 ) {
Merged revisions 264996 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines
Allow ast_safe_sleep to defer specific frames until after the sleep has concluded.
From reviewboard
Background:
A Digium customer discovered a somewhat odd bug. The setup is that parties A
and B are bridged, and party A places party B on hold. While party B is
listening to hold music, he mashes a bunch of DTMF. Party A takes party
B off hold while this is happening, but party B continues to hear hold
music. I could reproduce this about 1 in 5 times.
The issue:
When DTMF features are enabled and a user presses keys, the channel that
the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the
duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read
from the channel during the sleep, the frame is dropped. Thus the
unhold indication is never made to the channel that was originally placed
on hold.
The fix:
Originally, I discussed with Kevin possible ways of fixing the specific
problem reported. However, we determined that the same type of problem
could happen in other situations where ast_safe_sleep() is used. Using
autoservice as a model, I modified ast_safe_sleep_conditional() to
defer specific frame types so they can be re-queued once the sleep has
finished. I made a common function for determining if a frame should
be deferred so that there are not two identical switch blocks to
maintain.
Review: https://reviewboard.asterisk.org/r/674/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-21 16:44:27 +00:00
struct ast_frame * dup_f = NULL ;
2010-01-13 16:31:14 +00:00
if ( cond & & ( ( * cond ) ( data ) = = 0 ) ) {
break ;
}
2003-02-05 21:18:09 +00:00
ms = ast_waitfor ( chan , ms ) ;
2010-01-13 16:31:14 +00:00
if ( ms < 0 ) {
res = - 1 ;
break ;
}
2003-02-05 21:18:09 +00:00
if ( ms > 0 ) {
f = ast_read ( chan ) ;
2010-01-13 16:31:14 +00:00
if ( ! f ) {
res = - 1 ;
break ;
}
Merged revisions 264996 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines
Allow ast_safe_sleep to defer specific frames until after the sleep has concluded.
From reviewboard
Background:
A Digium customer discovered a somewhat odd bug. The setup is that parties A
and B are bridged, and party A places party B on hold. While party B is
listening to hold music, he mashes a bunch of DTMF. Party A takes party
B off hold while this is happening, but party B continues to hear hold
music. I could reproduce this about 1 in 5 times.
The issue:
When DTMF features are enabled and a user presses keys, the channel that
the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the
duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read
from the channel during the sleep, the frame is dropped. Thus the
unhold indication is never made to the channel that was originally placed
on hold.
The fix:
Originally, I discussed with Kevin possible ways of fixing the specific
problem reported. However, we determined that the same type of problem
could happen in other situations where ast_safe_sleep() is used. Using
autoservice as a model, I modified ast_safe_sleep_conditional() to
defer specific frame types so they can be re-queued once the sleep has
finished. I made a common function for determining if a frame should
be deferred so that there are not two identical switch blocks to
maintain.
Review: https://reviewboard.asterisk.org/r/674/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-21 16:44:27 +00:00
if ( ! ast_is_deferrable_frame ( f ) ) {
ast_frfree ( f ) ;
continue ;
}
if ( ( dup_f = ast_frisolate ( f ) ) ) {
if ( dup_f ! = f ) {
ast_frfree ( f ) ;
}
AST_LIST_INSERT_HEAD ( & deferred_frames , dup_f , frame_list ) ;
}
2003-02-05 21:18:09 +00:00
}
}
2010-01-13 16:31:14 +00:00
/* stop silgen if present */
if ( silgen ) {
ast_channel_stop_silence_generator ( chan , silgen ) ;
}
Merged revisions 264996 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines
Allow ast_safe_sleep to defer specific frames until after the sleep has concluded.
From reviewboard
Background:
A Digium customer discovered a somewhat odd bug. The setup is that parties A
and B are bridged, and party A places party B on hold. While party B is
listening to hold music, he mashes a bunch of DTMF. Party A takes party
B off hold while this is happening, but party B continues to hear hold
music. I could reproduce this about 1 in 5 times.
The issue:
When DTMF features are enabled and a user presses keys, the channel that
the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the
duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read
from the channel during the sleep, the frame is dropped. Thus the
unhold indication is never made to the channel that was originally placed
on hold.
The fix:
Originally, I discussed with Kevin possible ways of fixing the specific
problem reported. However, we determined that the same type of problem
could happen in other situations where ast_safe_sleep() is used. Using
autoservice as a model, I modified ast_safe_sleep_conditional() to
defer specific frame types so they can be re-queued once the sleep has
finished. I made a common function for determining if a frame should
be deferred so that there are not two identical switch blocks to
maintain.
Review: https://reviewboard.asterisk.org/r/674/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-21 16:44:27 +00:00
/* We need to free all the deferred frames, but we only need to
* queue the deferred frames if there was no error and no
* hangup was received
*/
ast_channel_lock ( chan ) ;
while ( ( f = AST_LIST_REMOVE_HEAD ( & deferred_frames , frame_list ) ) ) {
if ( ! res ) {
ast_queue_frame_head ( chan , f ) ;
}
ast_frfree ( f ) ;
}
ast_channel_unlock ( chan ) ;
2010-01-13 16:31:14 +00:00
return res ;
2003-02-05 21:18:09 +00:00
}
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
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
struct ast_channel * ast_channel_release ( struct ast_channel * chan )
{
/* Safe, even if already unlinked. */
ao2_unlink ( channels , chan ) ;
return ast_channel_unref ( chan ) ;
}
2010-07-14 15:48:36 +00:00
void ast_party_name_init ( struct ast_party_name * init )
{
init - > str = NULL ;
init - > char_set = AST_PARTY_CHAR_SET_ISO8859_1 ;
init - > presentation = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED ;
init - > valid = 0 ;
}
void ast_party_name_copy ( struct ast_party_name * dest , const struct ast_party_name * src )
{
if ( dest = = src ) {
/* Don't copy to self */
return ;
}
ast_free ( dest - > str ) ;
dest - > str = ast_strdup ( src - > str ) ;
dest - > char_set = src - > char_set ;
dest - > presentation = src - > presentation ;
dest - > valid = src - > valid ;
}
void ast_party_name_set_init ( struct ast_party_name * init , const struct ast_party_name * guide )
{
init - > str = NULL ;
init - > char_set = guide - > char_set ;
init - > presentation = guide - > presentation ;
init - > valid = guide - > valid ;
}
void ast_party_name_set ( struct ast_party_name * dest , const struct ast_party_name * src )
{
if ( dest = = src ) {
/* Don't set to self */
return ;
}
if ( src - > str & & src - > str ! = dest - > str ) {
ast_free ( dest - > str ) ;
dest - > str = ast_strdup ( src - > str ) ;
}
dest - > char_set = src - > char_set ;
dest - > presentation = src - > presentation ;
dest - > valid = src - > valid ;
}
void ast_party_name_free ( struct ast_party_name * doomed )
{
ast_free ( doomed - > str ) ;
doomed - > str = NULL ;
}
void ast_party_number_init ( struct ast_party_number * init )
{
init - > str = NULL ;
init - > plan = 0 ; /* Unknown */
init - > presentation = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED ;
init - > valid = 0 ;
}
void ast_party_number_copy ( struct ast_party_number * dest , const struct ast_party_number * src )
{
if ( dest = = src ) {
/* Don't copy to self */
return ;
}
ast_free ( dest - > str ) ;
dest - > str = ast_strdup ( src - > str ) ;
dest - > plan = src - > plan ;
dest - > presentation = src - > presentation ;
dest - > valid = src - > valid ;
}
void ast_party_number_set_init ( struct ast_party_number * init , const struct ast_party_number * guide )
{
init - > str = NULL ;
init - > plan = guide - > plan ;
init - > presentation = guide - > presentation ;
init - > valid = guide - > valid ;
}
void ast_party_number_set ( struct ast_party_number * dest , const struct ast_party_number * src )
{
if ( dest = = src ) {
/* Don't set to self */
return ;
}
if ( src - > str & & src - > str ! = dest - > str ) {
ast_free ( dest - > str ) ;
dest - > str = ast_strdup ( src - > str ) ;
}
dest - > plan = src - > plan ;
dest - > presentation = src - > presentation ;
dest - > valid = src - > valid ;
}
void ast_party_number_free ( struct ast_party_number * doomed )
{
ast_free ( doomed - > str ) ;
doomed - > str = NULL ;
}
2009-10-22 16:33:22 +00:00
void ast_party_subaddress_init ( struct ast_party_subaddress * init )
{
init - > str = NULL ;
init - > type = 0 ;
init - > odd_even_indicator = 0 ;
init - > valid = 0 ;
}
void ast_party_subaddress_copy ( struct ast_party_subaddress * dest , const struct ast_party_subaddress * src )
{
if ( dest = = src ) {
/* Don't copy to self */
return ;
}
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ast_free ( dest - > str ) ;
2009-10-22 16:33:22 +00:00
dest - > str = ast_strdup ( src - > str ) ;
dest - > type = src - > type ;
dest - > odd_even_indicator = src - > odd_even_indicator ;
dest - > valid = src - > valid ;
}
void ast_party_subaddress_set_init ( struct ast_party_subaddress * init , const struct ast_party_subaddress * guide )
{
init - > str = NULL ;
init - > type = guide - > type ;
init - > odd_even_indicator = guide - > odd_even_indicator ;
init - > valid = guide - > valid ;
}
void ast_party_subaddress_set ( struct ast_party_subaddress * dest , const struct ast_party_subaddress * src )
{
if ( dest = = src ) {
/* Don't set to self */
return ;
}
if ( src - > str & & src - > str ! = dest - > str ) {
2010-07-14 15:48:36 +00:00
ast_free ( dest - > str ) ;
2009-10-22 16:33:22 +00:00
dest - > str = ast_strdup ( src - > str ) ;
}
dest - > type = src - > type ;
dest - > odd_even_indicator = src - > odd_even_indicator ;
dest - > valid = src - > valid ;
}
void ast_party_subaddress_free ( struct ast_party_subaddress * doomed )
{
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ast_free ( doomed - > str ) ;
doomed - > str = NULL ;
2009-10-22 16:33:22 +00:00
}
2010-06-02 22:28:58 +00:00
void ast_party_id_init ( struct ast_party_id * init )
2009-04-03 22:41:46 +00:00
{
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ast_party_name_init ( & init - > name ) ;
ast_party_number_init ( & init - > number ) ;
2009-10-22 16:33:22 +00:00
ast_party_subaddress_init ( & init - > subaddress ) ;
2010-07-14 15:48:36 +00:00
init - > tag = NULL ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
void ast_party_id_copy ( struct ast_party_id * dest , const struct ast_party_id * src )
2009-04-03 22:41:46 +00:00
{
if ( dest = = src ) {
/* Don't copy to self */
return ;
}
2010-07-14 15:48:36 +00:00
ast_party_name_copy ( & dest - > name , & src - > name ) ;
ast_party_number_copy ( & dest - > number , & src - > number ) ;
ast_party_subaddress_copy ( & dest - > subaddress , & src - > subaddress ) ;
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
ast_free ( dest - > tag ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
dest - > tag = ast_strdup ( src - > tag ) ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
void ast_party_id_set_init ( struct ast_party_id * init , const struct ast_party_id * guide )
2009-04-03 22:41:46 +00:00
{
2010-07-14 15:48:36 +00:00
ast_party_name_set_init ( & init - > name , & guide - > name ) ;
ast_party_number_set_init ( & init - > number , & guide - > number ) ;
2009-10-22 16:33:22 +00:00
ast_party_subaddress_set_init ( & init - > subaddress , & guide - > subaddress ) ;
2010-07-14 15:48:36 +00:00
init - > tag = NULL ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
void ast_party_id_set ( struct ast_party_id * dest , const struct ast_party_id * src , const struct ast_set_party_id * update )
2009-04-03 22:41:46 +00:00
{
if ( dest = = src ) {
/* Don't set to self */
return ;
}
2010-07-14 15:48:36 +00:00
if ( ! update | | update - > name ) {
ast_party_name_set ( & dest - > name , & src - > name ) ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
if ( ! update | | update - > number ) {
ast_party_number_set ( & dest - > number , & src - > number ) ;
}
if ( ! update | | update - > subaddress ) {
ast_party_subaddress_set ( & dest - > subaddress , & src - > subaddress ) ;
2009-04-03 22:41:46 +00:00
}
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
if ( src - > tag & & src - > tag ! = dest - > tag ) {
2010-07-14 15:48:36 +00:00
ast_free ( dest - > tag ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
dest - > tag = ast_strdup ( src - > tag ) ;
}
2009-04-03 22:41:46 +00:00
}
2010-06-02 22:28:58 +00:00
void ast_party_id_free ( struct ast_party_id * doomed )
2009-04-03 22:41:46 +00:00
{
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ast_party_name_free ( & doomed - > name ) ;
ast_party_number_free ( & doomed - > number ) ;
ast_party_subaddress_free ( & doomed - > subaddress ) ;
ast_free ( doomed - > tag ) ;
doomed - > tag = NULL ;
}
int ast_party_id_presentation ( const struct ast_party_id * id )
{
int number_priority ;
int number_value ;
int number_screening ;
int name_priority ;
int name_value ;
/* Determine name presentation priority. */
if ( ! id - > name . valid ) {
name_value = AST_PRES_UNAVAILABLE ;
name_priority = 3 ;
} else {
name_value = id - > name . presentation & AST_PRES_RESTRICTION ;
switch ( name_value ) {
case AST_PRES_RESTRICTED :
name_priority = 0 ;
break ;
case AST_PRES_ALLOWED :
name_priority = 1 ;
break ;
case AST_PRES_UNAVAILABLE :
name_priority = 2 ;
break ;
default :
name_value = AST_PRES_UNAVAILABLE ;
name_priority = 3 ;
break ;
}
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
/* Determine number presentation priority. */
if ( ! id - > number . valid ) {
number_screening = AST_PRES_USER_NUMBER_UNSCREENED ;
number_value = AST_PRES_UNAVAILABLE ;
number_priority = 3 ;
} else {
number_screening = id - > number . presentation & AST_PRES_NUMBER_TYPE ;
number_value = id - > number . presentation & AST_PRES_RESTRICTION ;
switch ( number_value ) {
case AST_PRES_RESTRICTED :
number_priority = 0 ;
break ;
case AST_PRES_ALLOWED :
number_priority = 1 ;
break ;
case AST_PRES_UNAVAILABLE :
number_priority = 2 ;
break ;
default :
number_screening = AST_PRES_USER_NUMBER_UNSCREENED ;
number_value = AST_PRES_UNAVAILABLE ;
number_priority = 3 ;
break ;
}
2009-04-03 22:41:46 +00:00
}
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
2010-07-14 15:48:36 +00:00
/* Select the wining presentation value. */
if ( name_priority < number_priority ) {
number_value = name_value ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
}
2010-07-14 15:48:36 +00:00
return number_value | number_screening ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
void ast_party_dialed_init ( struct ast_party_dialed * init )
2009-08-04 16:36:41 +00:00
{
2010-07-14 15:48:36 +00:00
init - > number . str = NULL ;
init - > number . plan = 0 ; /* Unknown */
ast_party_subaddress_init ( & init - > subaddress ) ;
init - > transit_network_select = 0 ;
2009-08-04 16:36:41 +00:00
}
2010-07-14 15:48:36 +00:00
void ast_party_dialed_copy ( struct ast_party_dialed * dest , const struct ast_party_dialed * src )
2009-04-03 22:41:46 +00:00
{
if ( dest = = src ) {
/* Don't copy to self */
return ;
}
2010-07-14 15:48:36 +00:00
ast_free ( dest - > number . str ) ;
dest - > number . str = ast_strdup ( src - > number . str ) ;
dest - > number . plan = src - > number . plan ;
ast_party_subaddress_copy ( & dest - > subaddress , & src - > subaddress ) ;
dest - > transit_network_select = src - > transit_network_select ;
}
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
void ast_party_dialed_set_init ( struct ast_party_dialed * init , const struct ast_party_dialed * guide )
{
init - > number . str = NULL ;
init - > number . plan = guide - > number . plan ;
ast_party_subaddress_set_init ( & init - > subaddress , & guide - > subaddress ) ;
init - > transit_network_select = guide - > transit_network_select ;
}
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
void ast_party_dialed_set ( struct ast_party_dialed * dest , const struct ast_party_dialed * src )
{
if ( src - > number . str & & src - > number . str ! = dest - > number . str ) {
ast_free ( dest - > number . str ) ;
dest - > number . str = ast_strdup ( src - > number . str ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
}
2010-07-14 15:48:36 +00:00
dest - > number . plan = src - > number . plan ;
ast_party_subaddress_set ( & dest - > subaddress , & src - > subaddress ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
2010-07-14 15:48:36 +00:00
dest - > transit_network_select = src - > transit_network_select ;
}
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
void ast_party_dialed_free ( struct ast_party_dialed * doomed )
{
ast_free ( doomed - > number . str ) ;
doomed - > number . str = NULL ;
ast_party_subaddress_free ( & doomed - > subaddress ) ;
}
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
void ast_party_caller_init ( struct ast_party_caller * init )
{
ast_party_id_init ( & init - > id ) ;
2010-07-14 16:58:03 +00:00
ast_party_id_init ( & init - > ani ) ;
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init - > ani2 = 0 ;
}
void ast_party_caller_copy ( struct ast_party_caller * dest , const struct ast_party_caller * src )
{
if ( dest = = src ) {
/* Don't copy to self */
return ;
2009-04-03 22:41:46 +00:00
}
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ast_party_id_copy ( & dest - > id , & src - > id ) ;
2010-07-14 16:58:03 +00:00
ast_party_id_copy ( & dest - > ani , & src - > ani ) ;
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dest - > ani2 = src - > ani2 ;
}
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
void ast_party_caller_set_init ( struct ast_party_caller * init , const struct ast_party_caller * guide )
{
ast_party_id_set_init ( & init - > id , & guide - > id ) ;
2010-07-14 16:58:03 +00:00
ast_party_id_set_init ( & init - > ani , & guide - > ani ) ;
2010-07-14 15:48:36 +00:00
init - > ani2 = guide - > ani2 ;
}
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
void ast_party_caller_set ( struct ast_party_caller * dest , const struct ast_party_caller * src , const struct ast_set_party_caller * update )
{
ast_party_id_set ( & dest - > id , & src - > id , update ? & update - > id : NULL ) ;
2010-07-14 16:58:03 +00:00
ast_party_id_set ( & dest - > ani , & src - > ani , update ? & update - > ani : NULL ) ;
2009-04-03 22:41:46 +00:00
dest - > ani2 = src - > ani2 ;
2010-07-14 15:48:36 +00:00
}
void ast_party_caller_free ( struct ast_party_caller * doomed )
{
ast_party_id_free ( & doomed - > id ) ;
2010-07-14 16:58:03 +00:00
ast_party_id_free ( & doomed - > ani ) ;
2009-04-03 22:41:46 +00:00
}
void ast_party_connected_line_init ( struct ast_party_connected_line * init )
{
ast_party_id_init ( & init - > id ) ;
2010-07-14 16:58:03 +00:00
ast_party_id_init ( & init - > ani ) ;
2009-04-03 22:41:46 +00:00
init - > ani2 = 0 ;
init - > source = AST_CONNECTED_LINE_UPDATE_SOURCE_UNKNOWN ;
}
void ast_party_connected_line_copy ( struct ast_party_connected_line * dest , const struct ast_party_connected_line * src )
{
if ( dest = = src ) {
/* Don't copy to self */
return ;
}
ast_party_id_copy ( & dest - > id , & src - > id ) ;
2010-07-14 16:58:03 +00:00
ast_party_id_copy ( & dest - > ani , & src - > ani ) ;
2009-04-03 22:41:46 +00:00
dest - > ani2 = src - > ani2 ;
dest - > source = src - > source ;
}
void ast_party_connected_line_set_init ( struct ast_party_connected_line * init , const struct ast_party_connected_line * guide )
{
ast_party_id_set_init ( & init - > id , & guide - > id ) ;
2010-07-14 16:58:03 +00:00
ast_party_id_set_init ( & init - > ani , & guide - > ani ) ;
2009-04-03 22:41:46 +00:00
init - > ani2 = guide - > ani2 ;
init - > source = guide - > source ;
}
2010-07-14 15:48:36 +00:00
void ast_party_connected_line_set ( struct ast_party_connected_line * dest , const struct ast_party_connected_line * src , const struct ast_set_party_connected_line * update )
2009-04-03 22:41:46 +00:00
{
2010-07-14 15:48:36 +00:00
ast_party_id_set ( & dest - > id , & src - > id , update ? & update - > id : NULL ) ;
2010-07-14 16:58:03 +00:00
ast_party_id_set ( & dest - > ani , & src - > ani , update ? & update - > ani : NULL ) ;
2009-04-03 22:41:46 +00:00
dest - > ani2 = src - > ani2 ;
dest - > source = src - > source ;
}
2010-07-14 15:48:36 +00:00
void ast_party_connected_line_collect_caller ( struct ast_party_connected_line * connected , struct ast_party_caller * caller )
2009-04-03 22:41:46 +00:00
{
2010-07-14 15:48:36 +00:00
connected - > id = caller - > id ;
connected - > ani = caller - > ani ;
connected - > ani2 = caller - > ani2 ;
2009-04-03 22:41:46 +00:00
connected - > source = AST_CONNECTED_LINE_UPDATE_SOURCE_UNKNOWN ;
}
void ast_party_connected_line_free ( struct ast_party_connected_line * doomed )
{
ast_party_id_free ( & doomed - > id ) ;
2010-07-14 16:58:03 +00:00
ast_party_id_free ( & doomed - > ani ) ;
2010-07-14 15:48:36 +00:00
}
void ast_party_redirecting_init ( struct ast_party_redirecting * init )
{
ast_party_id_init ( & init - > from ) ;
ast_party_id_init ( & init - > to ) ;
init - > count = 0 ;
init - > reason = AST_REDIRECTING_REASON_UNKNOWN ;
2009-04-03 22:41:46 +00:00
}
void ast_party_redirecting_copy ( struct ast_party_redirecting * dest , const struct ast_party_redirecting * src )
{
if ( dest = = src ) {
/* Don't copy to self */
return ;
}
ast_party_id_copy ( & dest - > from , & src - > from ) ;
ast_party_id_copy ( & dest - > to , & src - > to ) ;
dest - > count = src - > count ;
dest - > reason = src - > reason ;
}
void ast_party_redirecting_set_init ( struct ast_party_redirecting * init , const struct ast_party_redirecting * guide )
{
ast_party_id_set_init ( & init - > from , & guide - > from ) ;
ast_party_id_set_init ( & init - > to , & guide - > to ) ;
init - > count = guide - > count ;
init - > reason = guide - > reason ;
}
2010-07-14 15:48:36 +00:00
void ast_party_redirecting_set ( struct ast_party_redirecting * dest , const struct ast_party_redirecting * src , const struct ast_set_party_redirecting * update )
{
ast_party_id_set ( & dest - > from , & src - > from , update ? & update - > from : NULL ) ;
ast_party_id_set ( & dest - > to , & src - > to , update ? & update - > to : NULL ) ;
dest - > reason = src - > reason ;
dest - > count = src - > count ;
}
2009-04-03 22:41:46 +00:00
void ast_party_redirecting_free ( struct ast_party_redirecting * doomed )
{
ast_party_id_free ( & doomed - > from ) ;
ast_party_id_free ( & doomed - > to ) ;
}
2005-12-31 16:35:48 +00:00
/*! \brief Free a channel structure */
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
static void ast_channel_destructor ( void * obj )
1999-12-18 07:01:48 +00:00
{
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
struct ast_channel * chan = obj ;
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 ;
2011-03-14 21:30:25 +00:00
struct ast_datastore * datastore ;
2010-05-21 22:46:52 +00:00
char device_name [ AST_CHANNEL_NAME ] ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
2010-05-21 22:46:52 +00:00
if ( chan - > name ) {
/* The string fields were initialized. */
ast_cel_report_event ( chan , AST_CEL_CHANNEL_END , NULL , NULL , NULL ) ;
ast_cel_check_retire_linkedid ( chan ) ;
}
2009-06-26 15:28:53 +00:00
2011-03-14 21:30:25 +00:00
/* Get rid of each of the data stores on the channel */
ast_channel_lock ( chan ) ;
while ( ( datastore = AST_LIST_REMOVE_HEAD ( & chan - > datastores , entry ) ) )
/* Free the data store */
ast_datastore_free ( datastore ) ;
ast_channel_unlock ( chan ) ;
/* 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: https://origsvn.digium.com/svn/asterisk/trunk@7276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-02 01:01:11 +00:00
if ( chan - > sched )
sched_context_destroy ( chan - > sched ) ;
2010-05-21 22:46:52 +00:00
if ( chan - > name ) {
char * dashptr ;
/* The string fields were initialized. */
ast_copy_string ( device_name , chan - > name , sizeof ( device_name ) ) ;
if ( ( dashptr = strrchr ( device_name , ' - ' ) ) ) {
* dashptr = ' \0 ' ;
}
} else {
device_name [ 0 ] = ' \0 ' ;
2008-07-25 17:24:43 +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: https://origsvn.digium.com/svn/asterisk/trunk@7276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 ) ;
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
ast_party_dialed_free ( & chan - > dialed ) ;
ast_party_caller_free ( & chan - > caller ) ;
2009-04-03 22:41:46 +00:00
ast_party_connected_line_free ( & chan - > connected ) ;
ast_party_redirecting_free ( & chan - > redirecting ) ;
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 ) ;
2009-03-27 19:10:32 +00:00
if ( chan - > timer ) {
ast_timer_close ( chan - > timer ) ;
}
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 ) ;
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/* loop over the variables list, freeing all data and deleting list items */
/* no need to lock the list, as the channel is already locked */
2010-05-21 22:46:52 +00:00
headp = & chan - > varshead ;
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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 ) ;
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/* Destroy the jitterbuffer */
2006-06-23 16:49:12 +00:00
ast_jb_destroy ( chan ) ;
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if ( chan - > cdr ) {
2008-09-05 14:18:43 +00:00
ast_cdr_discard ( chan - > cdr ) ;
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chan - > cdr = NULL ;
}
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if ( chan - > zone ) {
chan - > zone = ast_tone_zone_unref ( chan - > zone ) ;
}
2007-11-04 19:44:31 +00:00
ast_string_field_free_memory ( chan ) ;
2003-03-30 22:55:42 +00:00
2010-05-21 22:46:52 +00:00
if ( device_name [ 0 ] ) {
/*
* We have a device name to notify of a new state .
*
* Queue an unknown state , because , while we know that this particular
* instance is dead , we don ' t know the state of all other possible
* instances .
*/
ast_devstate_changed_literal ( AST_DEVICE_UNKNOWN , device_name ) ;
}
1999-12-18 07:01:48 +00:00
}
2009-06-26 15:28:53 +00:00
/*! \brief Free a dummy channel structure */
static void ast_dummy_channel_destructor ( void * obj )
{
struct ast_channel * chan = obj ;
struct ast_var_t * vardata ;
struct varshead * headp ;
headp = & chan - > varshead ;
2010-07-14 15:48:36 +00:00
ast_party_dialed_free ( & chan - > dialed ) ;
ast_party_caller_free ( & chan - > caller ) ;
ast_party_connected_line_free ( & chan - > connected ) ;
ast_party_redirecting_free ( & chan - > redirecting ) ;
2009-06-26 15:28:53 +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 */
while ( ( vardata = AST_LIST_REMOVE_HEAD ( headp , entries ) ) )
ast_var_delete ( vardata ) ;
if ( chan - > cdr ) {
ast_cdr_discard ( chan - > cdr ) ;
chan - > cdr = NULL ;
}
ast_string_field_free_memory ( chan ) ;
}
2007-08-28 18:32:56 +00:00
struct ast_datastore * ast_channel_datastore_alloc ( const struct ast_datastore_info * info , const char * uid )
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{
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return ast_datastore_alloc ( info , uid ) ;
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}
int ast_channel_datastore_free ( struct ast_datastore * datastore )
{
2008-08-05 16:56:11 +00:00
return ast_datastore_free ( datastore ) ;
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}
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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 ) ;
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if ( datastore2 ) {
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datastore2 - > data = datastore - > info - > duplicate ? datastore - > info - > duplicate ( datastore - > data ) : NULL ;
2007-03-03 14:40:18 +00:00
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 ;
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AST_LIST_INSERT_HEAD ( & chan - > datastores , datastore , entry ) ;
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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 ;
2011-05-03 19:55:49 +00:00
AST_LIST_TRAVERSE ( & 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
}
}
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 ;
}
Merged revisions 303548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
Merged revisions 303546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
Fix channel redirect out of MeetMe() and other issues with channel softhangup.
Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
working properly. This issue includes a patch that resolves the issue by
removing a call to ast_check_hangup() from app_meetme.c. I left that in my
patch, as it doesn't need to be there. However, the rest of the patch fixes
this problem with or without the change to app_meetme.
The key difference between what happens before and after this patch is the
effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(),
ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme
sees this which causes it to exit as intended. Checking ast_check_hangup()
caused app_meetme to exit earlier in the process, and the target of the
redirect saw the condition where ast_read() returned NULL.
Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
solve the issue if another application did the same thing. There are also
other edge cases where if an application finishes at the same time that a
redirect happens, the target of the redirect will think that the channel hung
up. So, I made some changes in pbx.c to resolve it at a deeper level. There
are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
abort the hangup process. My patch extends this to remove the END_OF_Q frame
from the channel's read queue, making the "abort hangup" more complete. This
same technique was used in every place where a softhangup flag was cleared.
(closes issue #18585)
Reported by: oej
Tested by: oej, wedhorn, russell
Review: https://reviewboard.asterisk.org/r/1082/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-24 20:51:37 +00:00
void ast_channel_clear_softhangup ( struct ast_channel * chan , int flag )
{
ast_channel_lock ( chan ) ;
chan - > _softhangup & = ~ flag ;
if ( ! chan - > _softhangup ) {
struct ast_frame * fr ;
/* If we have completely cleared the softhangup flag,
* then we need to fully abort the hangup process . This requires
* pulling the END_OF_Q frame out of the channel frame queue if it
* still happens to be there . */
fr = AST_LIST_LAST ( & chan - > readq ) ;
if ( fr & & fr - > frametype = = AST_FRAME_CONTROL & &
fr - > subclass . integer = = AST_CONTROL_END_OF_Q ) {
AST_LIST_REMOVE ( & chan - > readq , fr , frame_list ) ;
ast_frfree ( fr ) ;
}
}
ast_channel_unlock ( chan ) ;
}
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 */
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chan - > _softhangup | = cause ;
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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
}
2009-06-26 15:28:53 +00:00
void ast_set_hangupsource ( struct ast_channel * chan , const char * source , int force )
{
struct ast_channel * bridge ;
ast_channel_lock ( chan ) ;
if ( force | | ast_strlen_zero ( chan - > hangupsource ) ) {
ast_string_field_set ( chan , hangupsource , source ) ;
}
bridge = ast_bridged_channel ( chan ) ;
ast_channel_unlock ( chan ) ;
if ( bridge & & ( force | | ast_strlen_zero ( bridge - > hangupsource ) ) ) {
ast_channel_lock ( bridge ) ;
ast_string_field_set ( chan , hangupsource , source ) ;
ast_channel_unlock ( bridge ) ;
}
}
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 )
{
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char extra_str [ 64 ] ; /* used for cel logging below */
2005-07-25 22:13:32 +00:00
2011-08-31 15:20:31 +00:00
ast_autoservice_stop ( chan ) ;
ao2_lock ( channels ) ;
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 ;
}
2010-09-20 22:09:16 +00:00
ast_framehook_list_destroy ( chan ) ;
2011-08-31 15:20:31 +00:00
/*
* Do the masquerade if someone is setup to masquerade into us .
*
* NOTE : We must hold the channel lock after testing for a
* pending masquerade and setting the channel as a zombie to
* prevent __ast_channel_masquerade ( ) from setting up a
* masquerade with a dead channel .
*/
while ( chan - > masq ) {
2009-10-07 22:58:38 +00:00
ast_channel_unlock ( chan ) ;
2011-08-31 15:20:31 +00:00
ao2_unlock ( channels ) ;
2009-10-07 22:58:38 +00:00
if ( ast_do_masquerade ( chan ) ) {
2003-02-05 21:18:09 +00:00
ast_log ( LOG_WARNING , " Failed to perform masquerade \n " ) ;
2011-08-31 15:20:31 +00:00
/* Abort the loop or we might never leave. */
ao2_lock ( channels ) ;
ast_channel_lock ( chan ) ;
break ;
2009-10-07 22:58:38 +00:00
}
2011-08-31 15:20:31 +00:00
ao2_lock ( channels ) ;
2009-10-07 22:58:38 +00:00
ast_channel_lock ( chan ) ;
2003-02-05 21:18:09 +00:00
}
2001-05-07 00:43:32 +00:00
if ( chan - > masqr ) {
2011-08-31 15:20:31 +00:00
/*
* This channel is one which will be masqueraded into something .
* Mark it as a zombie already so ast_do_masquerade ( ) will know
* to free it later .
*/
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 ) ;
2011-08-31 15:20:31 +00:00
ao2_unlock ( channels ) ;
2001-05-07 00:43:32 +00:00
return 0 ;
}
2009-09-17 15:18:01 +00:00
ao2_unlink ( channels , chan ) ;
2011-08-31 15:20:31 +00:00
ao2_unlock ( channels ) ;
2009-09-17 15:18:01 +00:00
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: https://origsvn.digium.com/svn/asterisk/trunk@7276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@7276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-02 01:01:11 +00:00
chan - > sched = NULL ;
}
2009-06-26 15:28:53 +00:00
2011-08-31 15:20:31 +00:00
if ( chan - > generatordata ) { /* Clear any tone stuff remaining */
if ( chan - > generator & & chan - > generator - > release ) {
2006-12-21 19:36:42 +00:00
chan - > generator - > release ( chan , chan - > generatordata ) ;
2011-08-31 15:20:31 +00:00
}
}
2002-06-21 01:40:13 +00:00
chan - > generatordata = NULL ;
chan - > generator = NULL ;
2009-06-26 15:28:53 +00:00
snprintf ( extra_str , sizeof ( extra_str ) , " %d,%s,%s " , chan - > hangupcause , chan - > hangupsource , S_OR ( pbx_builtin_getvar_helper ( chan , " DIALSTATUS " ) , " " ) ) ;
ast_cel_report_event ( chan , AST_CEL_HANGUP , NULL , extra_str , 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 "
2011-08-31 15:20:31 +00:00
" is blocked by thread %ld in procedure %s! Expect a failure \n " ,
( 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 ) ;
2011-08-31 15:20:31 +00:00
/*
* This channel is now dead so mark it as a zombie so anyone
* left holding a reference to this channel will not use it .
*/
ast_set_flag ( chan , AST_FLAG_ZOMBIE ) ;
if ( chan - > tech - > hangup ) {
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
}
2011-08-31 15:20:31 +00:00
2006-04-15 15:07:39 +00:00
ast_channel_unlock ( chan ) ;
2011-08-31 15:20:31 +00:00
Merge Call completion support into trunk.
From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:
1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
multiple calls to the same device. This proved to not be such a good idea
when implementing protocol-specific monitors, and so we ended up using one
monitor per-device per-call.
3. There are some configuration options which were conceived after the document
was written. These are documented in the ccss.conf.sample that is on this
review request.
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.
This implements CCBS and CCNR in several flavors.
First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.
Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:
* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
what is defined in the referenced draft.
* Implementation of the draft required support for SIP PUBLISH. I attempted to write
this in a generic-enough fashion such that if someone were to want to write PUBLISH
support for other event packages, such as dialog-state or presence, most of the effort
would be in writing callbacks specific to the event package.
* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
parser. The PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly. The rest of the
PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
code. In other words, while there is PIDF support here, it is not in any state
where it could easily be applied to other event packages as is.
Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.
Review: https://reviewboard.asterisk.org/r/523
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09 15:31:32 +00:00
ast_cc_offer ( chan ) ;
2009-11-13 20:42:03 +00:00
ast_manager_event ( chan , EVENT_FLAG_CALL , " Hangup " ,
2010-07-14 15:48:36 +00:00
" Channel: %s \r \n "
" Uniqueid: %s \r \n "
" CallerIDNum: %s \r \n "
" CallerIDName: %s \r \n "
2011-05-25 17:06:38 +00:00
" ConnectedLineNum: %s \r \n "
" ConnectedLineName: %s \r \n "
2010-07-14 15:48:36 +00:00
" Cause: %d \r \n "
" Cause-txt: %s \r \n " ,
chan - > name ,
chan - > uniqueid ,
S_COR ( chan - > caller . id . number . valid , chan - > caller . id . number . str , " <unknown> " ) ,
S_COR ( chan - > caller . id . name . valid , chan - > caller . id . name . str , " <unknown> " ) ,
2011-05-25 17:06:38 +00:00
S_COR ( chan - > connected . id . number . valid , chan - > connected . id . number . str , " <unknown> " ) ,
S_COR ( chan - > connected . id . name . valid , chan - > connected . id . name . str , " <unknown> " ) ,
2010-07-14 15:48:36 +00:00
chan - > hangupcause ,
ast_cause2str ( chan - > hangupcause )
) ;
2007-06-27 20:26:53 +00:00
2011-08-31 15:20:31 +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 ) ) ) {
2009-08-06 21:46:01 +00:00
ast_channel_lock ( chan ) ;
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 ;
2009-08-06 21:46:01 +00:00
ast_channel_unlock ( chan ) ;
2008-07-03 17:16:44 +00:00
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
2011-08-31 15:20:31 +00:00
ast_channel_unref ( chan ) ;
2007-06-27 20:26:53 +00:00
2011-08-31 15:20:31 +00:00
return 0 ;
1999-12-04 21:35:07 +00:00
}
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17 14:38:11 +00:00
int ast_raw_answer ( struct ast_channel * chan , int cdr_answer )
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 ) ;
2009-02-11 22:41:01 +00:00
if ( cdr_answer ) {
ast_cdr_answer ( chan - > cdr ) ;
}
2009-06-26 15:28:53 +00:00
ast_cel_report_event ( chan , AST_CEL_ANSWER , NULL , NULL , NULL ) ;
2008-09-09 01:47:56 +00:00
ast_channel_unlock ( chan ) ;
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17 14:38:11 +00:00
break ;
case AST_STATE_UP :
2009-06-26 15:28:53 +00:00
ast_cel_report_event ( chan , AST_CEL_ANSWER , NULL , NULL , NULL ) ;
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17 14:38:11 +00:00
/* Calling ast_cdr_answer when it it has previously been called
* is essentially a no - op , so it is safe .
*/
if ( cdr_answer ) {
ast_cdr_answer ( chan - > cdr ) ;
}
break ;
default :
break ;
}
ast_indicate ( chan , - 1 ) ;
return res ;
}
int __ast_answer ( struct ast_channel * chan , unsigned int delay , int cdr_answer )
{
int res = 0 ;
enum ast_channel_state old_state ;
old_state = chan - > _state ;
if ( ( res = ast_raw_answer ( chan , cdr_answer ) ) ) {
return res ;
}
switch ( old_state ) {
case AST_STATE_RINGING :
case AST_STATE_RING :
/* wait for media to start flowing, but don't wait any longer
* than ' delay ' or 500 milliseconds , whichever is longer
*/
do {
AST_LIST_HEAD_NOLOCK ( , ast_frame ) frames ;
struct ast_frame * cur , * new ;
int ms = MAX ( delay , 500 ) ;
unsigned int done = 0 ;
AST_LIST_HEAD_INIT_NOLOCK ( & frames ) ;
for ( ; ; ) {
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 ) {
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17 14:38:11 +00:00
ast_debug ( 2 , " Didn't receive a media frame from %s within %d ms of answering. Continuing anyway \n " , chan - > name , MAX ( delay , 500 ) ) ;
2008-08-07 19:36:46 +00:00
break ;
}
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17 14:38:11 +00:00
cur = ast_read ( chan ) ;
if ( ! cur | | ( ( cur - > frametype = = AST_FRAME_CONTROL ) & &
2009-11-04 14:05:12 +00:00
( cur - > subclass . integer = = AST_CONTROL_HANGUP ) ) ) {
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17 14:38:11 +00:00
if ( cur ) {
ast_frfree ( cur ) ;
2009-03-16 13:58:24 +00:00
}
2008-08-07 19:36:46 +00:00
res = - 1 ;
ast_debug ( 2 , " Hangup of channel %s detected in answer routine \n " , chan - > name ) ;
break ;
}
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17 14:38:11 +00:00
if ( ( new = ast_frisolate ( cur ) ) ! = cur ) {
ast_frfree ( cur ) ;
}
2009-03-17 15:22:12 +00:00
AST_LIST_INSERT_HEAD ( & frames , new , frame_list ) ;
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17 14:38:11 +00:00
/* if a specific delay period was requested, continue
* until that delay has passed . don ' t stop just because
* incoming media has arrived .
*/
if ( delay ) {
continue ;
}
switch ( new - > frametype ) {
/* all of these frametypes qualify as 'media' */
case AST_FRAME_VOICE :
case AST_FRAME_VIDEO :
case AST_FRAME_TEXT :
case AST_FRAME_DTMF_BEGIN :
case AST_FRAME_DTMF_END :
case AST_FRAME_IMAGE :
case AST_FRAME_HTML :
case AST_FRAME_MODEM :
done = 1 ;
break ;
case AST_FRAME_CONTROL :
case AST_FRAME_IAX :
case AST_FRAME_NULL :
case AST_FRAME_CNG :
break ;
}
if ( done ) {
2008-08-07 19:36:46 +00:00
break ;
}
2008-07-01 19:53:03 +00:00
}
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17 14:38:11 +00:00
if ( res = = 0 ) {
ast_channel_lock ( chan ) ;
2009-03-17 15:22:12 +00:00
while ( ( cur = AST_LIST_REMOVE_HEAD ( & frames , frame_list ) ) ) {
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17 14:38:11 +00:00
ast_queue_frame_head ( chan , cur ) ;
ast_frfree ( cur ) ;
}
ast_channel_unlock ( chan ) ;
}
} while ( 0 ) ;
2001-05-07 00:43:32 +00:00
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
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 )
{
2009-02-11 22:41:01 +00:00
return __ast_answer ( chan , 0 , 1 ) ;
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 ) ;
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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
2007-09-21 14:40:10 +00:00
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 ) ;
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tmp = chan - > generatordata ;
chan - > generatordata = NULL ;
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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: https://origsvn.digium.com/svn/asterisk/trunk@110268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 ) ;
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 ;
}
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 ;
}
2006-04-15 15:07:39 +00:00
ast_channel_unlock ( chan ) ;
2005-10-28 23:01:13 +00:00
2010-04-28 21:20:03 +00:00
ast_prod ( chan ) ;
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 ;
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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: https://origsvn.digium.com/svn/asterisk/trunk@105564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@105564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@105564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 + + ) {
if ( c [ x ] - > masq & & ast_do_masquerade ( c [ x ] ) ) {
ast_log ( LOG_WARNING , " Masquerade failed \n " ) ;
* ms = - 1 ;
return NULL ;
2001-05-07 00:43:32 +00:00
}
2009-10-07 22:58:38 +00:00
ast_channel_lock ( c [ x ] ) ;
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 ;
2010-05-04 23:51:52 +00:00
/* INT_MAX, not LONG_MAX, because it matters on 64-bit */
if ( ! ast_tvzero ( whentohangup ) & & whentohangup . tv_sec < INT_MAX / 1000 ) {
2008-05-01 23:06:23 +00:00
rms = whentohangup . tv_sec * 1000 + whentohangup . tv_usec / 1000 ; /* timeout in milliseconds */
2010-05-04 23:51:52 +00:00
if ( * ms > = 0 & & * ms < rms ) { /* original *ms still smaller */
2006-01-30 03:13:33 +00:00
rms = * ms ;
2010-05-04 23:51:52 +00:00
}
} else if ( ! ast_tvzero ( whentohangup ) & & rms < 0 ) {
/* Tiny corner case... call would need to last >24 days */
rms = INT_MAX ;
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 ;
Merged revisions 182810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines
Fix cases where the internal poll() was not being used when it needed to be.
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-18 02:28:55 +00:00
res = ast_poll ( pfds , max , kbrms ) ;
2005-10-20 22:45:47 +00:00
if ( ! res )
rms - = kbrms ;
} while ( ! res & & ( rms > 0 ) ) ;
} else {
Merged revisions 182810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines
Fix cases where the internal poll() was not being used when it needed to be.
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-18 02:28:55 +00:00
res = ast_poll ( pfds , max , rms ) ;
2005-10-20 22:45:47 +00:00
}
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 ;
/* 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 ;
return NULL ;
}
2009-10-07 22:58:38 +00:00
ast_channel_lock ( chan ) ;
2007-08-08 21:44:58 +00:00
/* 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 + + ) {
if ( c [ i ] - > masq & & ast_do_masquerade ( c [ i ] ) ) {
ast_log ( LOG_WARNING , " Masquerade failed \n " ) ;
* ms = - 1 ;
return NULL ;
}
2009-10-07 22:58:38 +00:00
ast_channel_lock ( c [ i ] ) ;
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
2009-03-02 23:00:30 +00:00
ast_channel_lock ( c ) ;
2008-06-13 12:45:50 +00:00
if ( c - > timingfd = = - 1 ) {
2009-03-02 23:00:30 +00:00
ast_channel_unlock ( c ) ;
2008-06-13 12:45:50 +00:00
return - 1 ;
}
if ( ! func ) {
rate = 0 ;
data = NULL ;
2003-05-12 04:23:55 +00:00
}
2008-06-13 12:45:50 +00:00
2009-03-27 19:10:32 +00:00
if ( rate & & rate > ( max_rate = ast_timer_get_max_rate ( c - > timer ) ) ) {
2008-06-26 15:37:01 +00:00
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
2009-03-27 19:10:32 +00:00
res = ast_timer_set_rate ( c - > timer , real_rate ) ;
2008-06-13 12:45:50 +00:00
c - > timingfunc = func ;
c - > timingdata = data ;
2009-03-02 23:00:30 +00:00
ast_channel_unlock ( c ) ;
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 :
2009-11-04 14:05:12 +00:00
res = f - > subclass . integer ;
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 :
2009-11-04 14:05:12 +00:00
switch ( f - > subclass . integer ) {
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 :
2010-03-12 22:04:51 +00:00
case AST_CONTROL_SRCCHANGE :
2009-04-03 22:41:46 +00:00
case AST_CONTROL_CONNECTED_LINE :
case AST_CONTROL_REDIRECTING :
2011-10-14 16:33:28 +00:00
case AST_CONTROL_UPDATE_RTP_PEER :
2010-07-26 21:53:30 +00:00
case - 1 :
2004-04-12 16:25:34 +00:00
/* Unimportant */
break ;
default :
2009-11-04 14:05:12 +00:00
ast_log ( LOG_WARNING , " Unexpected control subclass '%d' \n " , f - > subclass . integer ) ;
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
}
2009-11-13 20:42:03 +00:00
static void send_dtmf_event ( struct ast_channel * chan , const char * direction , const char digit , const char * begin , const char * end )
2007-04-10 23:55:26 +00:00
{
2009-11-13 20:42:03 +00:00
ast_manager_event ( chan , EVENT_FLAG_DTMF ,
2007-04-10 23:55:26 +00:00
" 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 )
{
2009-09-30 16:56:42 +00:00
if ( chan - > generator & & chan - > generator - > generate & & chan - > generatordata & & ! ast_internal_timing_enabled ( chan ) ) {
2007-06-27 17:34:26 +00:00
void * tmp = chan - > generatordata ;
2009-09-30 16:56:42 +00:00
int ( * generate ) ( struct ast_channel * chan , void * tmp , int datalen , int samples ) = chan - > generator - > generate ;
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: https://origsvn.digium.com/svn/asterisk/trunk@110268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@110268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-20 17:41:22 +00:00
2009-11-04 14:05:12 +00:00
if ( f - > subclass . codec ! = chan - > writeformat ) {
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: https://origsvn.digium.com/svn/asterisk/trunk@110268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-20 17:41:22 +00:00
float factor ;
2009-11-04 14:05:12 +00:00
factor = ( ( float ) ast_format_rate ( chan - > writeformat ) ) / ( ( float ) ast_format_rate ( f - > subclass . codec ) ) ;
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: https://origsvn.digium.com/svn/asterisk/trunk@110268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-20 17:41:22 +00:00
samples = ( int ) ( ( ( float ) f - > samples ) * factor ) ;
} else {
samples = f - > samples ;
}
2008-04-14 15:01:36 +00:00
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
}
}
}
2008-12-12 13:55:30 +00:00
static inline void queue_dtmf_readq ( struct ast_channel * chan , struct ast_frame * f )
{
struct ast_frame * fr = & chan - > dtmff ;
fr - > frametype = AST_FRAME_DTMF_END ;
2009-11-04 14:05:12 +00:00
fr - > subclass . integer = f - > subclass . integer ;
2008-12-12 13:55:30 +00:00
fr - > len = f - > len ;
/* The only time this function will be called is for a frame that just came
* out of the channel driver . So , we want to stick it on the tail of the
* readq . */
ast_queue_frame ( chan , fr ) ;
}
/*!
* \ brief Determine whether or not we should ignore DTMF in the readq
*/
static inline int should_skip_dtmf ( struct ast_channel * chan )
{
if ( ast_test_flag ( chan , AST_FLAG_DEFER_DTMF | AST_FLAG_EMULATE_DTMF ) ) {
/* We're in the middle of emulating a digit, or DTMF has been
* explicitly deferred . Skip this digit , then . */
return 1 ;
}
if ( ! ast_tvzero ( chan - > dtmf_tv ) & &
ast_tvdiff_ms ( ast_tvnow ( ) , chan - > dtmf_tv ) < AST_MIN_DTMF_GAP ) {
/* We're not in the middle of a digit, but it hasn't been long enough
* since the last digit , so we ' ll have to skip DTMF for now . */
return 1 ;
}
return 0 ;
}
2010-02-16 17:07:41 +00:00
/*!
* \ brief calculates the number of samples to jump forward with in a monitor stream .
* \ note When using ast_seekstream ( ) with the read and write streams of a monitor ,
* the number of samples to seek forward must be of the same sample rate as the stream
* or else the jump will not be calculated correctly .
*
* \ retval number of samples to seek forward after rate conversion .
*/
static inline int calc_monitor_jump ( int samples , int sample_rate , int seek_rate )
{
int diff = sample_rate - seek_rate ;
if ( diff > 0 ) {
samples = samples / ( float ) ( sample_rate / seek_rate ) ;
} else if ( diff < 0 ) {
samples = samples * ( float ) ( seek_rate / sample_rate ) ;
}
return samples ;
}
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 ;
2009-10-07 22:58:38 +00:00
int 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
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 ;
2009-10-07 22:58:38 +00:00
return f ;
2001-05-07 00:43:32 +00:00
}
2009-10-07 22:58:38 +00:00
/* if here, no masq has happened, lock the channel and proceed */
ast_channel_lock ( chan ) ;
2009-04-20 14:05:27 +00:00
/* Stop if we're a zombie or need a soft hangup */
if ( ast_test_flag ( chan , AST_FLAG_ZOMBIE ) | | ast_check_hangup ( chan ) ) {
if ( chan - > generator )
ast_deactivate_generator ( chan ) ;
2010-10-13 23:01:56 +00:00
2010-11-22 19:36:10 +00:00
/*
* It is possible for chan - > _softhangup to be set and there
* still be control frames that need to be read . Instead of
* just going to ' done ' in the case of ast_check_hangup ( ) , we
* need to queue the end - of - Q frame so that it can mark the end
* of the read queue . If there are frames to be read ,
* ast_queue_control ( ) will be called repeatedly , but will only
* queue the first end - of - Q frame .
*/
if ( chan - > _softhangup ) {
ast_queue_control ( chan , AST_CONTROL_END_OF_Q ) ;
2010-10-13 23:01:56 +00:00
} else {
goto done ;
}
2011-04-13 16:31:50 +00:00
} else {
2009-04-08 15:27:41 +00:00
# ifdef AST_DEVMODE
2011-04-13 16:31:50 +00:00
/*
* The ast_waitfor ( ) code records which of the channel ' s file
* descriptors reported that data is available . In theory ,
* ast_read ( ) should only be called after ast_waitfor ( ) reports
* that a channel has data available for reading . However ,
* there still may be some edge cases throughout the code where
* ast_read ( ) is called improperly . This can potentially cause
* problems , so if this is a developer build , make a lot of
* noise if this happens so that it can be addressed .
*
* One of the potential problems is blocking on a dead channel .
*/
if ( chan - > fdno = = - 1 ) {
ast_log ( LOG_ERROR ,
" ast_read() on chan '%s' called with no recorded file descriptor. \n " ,
chan - > name ) ;
}
2009-07-20 16:36:15 +00:00
# endif
2011-04-13 16:31:50 +00:00
}
2009-07-20 16:36:15 +00:00
2005-03-31 03:00:37 +00:00
prestate = chan - > _state ;
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 ) {
2009-02-17 21:22:40 +00:00
enum ast_timer_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
2009-03-27 19:10:32 +00:00
res = ast_timer_get_event ( chan - > timer ) ;
2008-06-16 12:48:11 +00:00
switch ( res ) {
case AST_TIMING_EVENT_EXPIRED :
2009-03-27 19:10:32 +00:00
ast_timer_ack ( chan - > timer , 1 ) ;
2008-06-16 12:48:11 +00:00
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 ;
2009-03-03 16:47:28 +00:00
chan - > fdno = - 1 ;
2008-01-28 17:21:24 +00:00
ast_channel_unlock ( chan ) ;
func ( data ) ;
2004-03-19 04:23:20 +00:00
} else {
2009-03-27 19:10:32 +00:00
ast_timer_set_rate ( chan - > timer , 0 ) ;
2009-03-03 16:47:28 +00:00
chan - > fdno = - 1 ;
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 ) ) {
2009-03-27 19:10:32 +00:00
ast_timer_disable_continuous ( chan - > timer ) ;
2008-06-16 12:48:11 +00:00
}
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 ;
2009-03-03 16:47:28 +00:00
chan - > fdno = - 1 ;
2006-04-16 17:41:05 +00:00
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 ) ) {
2008-12-12 13:55:30 +00:00
int skip_dtmf = should_skip_dtmf ( chan ) ;
AST_LIST_TRAVERSE_SAFE_BEGIN ( & chan - > readq , f , frame_list ) {
/* We have to be picky about which frame we pull off of the readq because
* there are cases where we want to leave DTMF frames on the queue until
* some later time . */
if ( ( f - > frametype = = AST_FRAME_DTMF_BEGIN | | f - > frametype = = AST_FRAME_DTMF_END ) & & skip_dtmf ) {
continue ;
}
AST_LIST_REMOVE_CURRENT ( frame_list ) ;
break ;
}
2009-04-10 03:55:27 +00:00
AST_LIST_TRAVERSE_SAFE_END ;
2008-12-12 13:55:30 +00:00
if ( ! f ) {
/* There were no acceptable frames on the readq. */
f = & ast_null_frame ;
if ( chan - > alertpipe [ 0 ] > - 1 ) {
int poke = 0 ;
/* Restore the state of the alertpipe since we aren't ready for any
* of the frames in the readq . */
if ( write ( chan - > alertpipe [ 1 ] , & poke , sizeof ( poke ) ) ! = sizeof ( poke ) ) {
ast_log ( LOG_ERROR , " Failed to write to alertpipe: %s \n " , strerror ( errno ) ) ;
}
}
}
2010-11-22 19:36:10 +00:00
/* Interpret hangup and end-of-Q frames to return NULL */
2006-05-09 07:51:55 +00:00
/* XXX why not the same for frames from the channel ? */
2010-11-22 19:36:10 +00:00
if ( f - > frametype = = AST_FRAME_CONTROL ) {
switch ( f - > subclass . integer ) {
case AST_CONTROL_HANGUP :
chan - > _softhangup | = AST_SOFTHANGUP_DEV ;
cause = f - > data . uint32 ;
/* Fall through */
case AST_CONTROL_END_OF_Q :
ast_frfree ( f ) ;
f = NULL ;
break ;
default :
break ;
}
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 ) ;
2011-05-13 18:09:34 +00:00
} else if ( chan - > tech & & chan - > tech - > read )
2006-04-16 13:58:43 +00:00
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
}
2009-03-03 16:47:28 +00:00
/*
2009-03-03 14:40:04 +00:00
* Reset the recorded file descriptor that triggered this read so that we can
* easily detect when ast_read ( ) is called without properly using ast_waitfor ( ) .
*/
chan - > fdno = - 1 ;
2010-09-20 22:09:16 +00:00
/* Perform the framehook read event here. After the frame enters the framehook list
* there is no telling what will happen , < insert mad scientist laugh here > ! ! ! */
f = ast_framehook_list_read_event ( chan - > framehooks , f ) ;
2006-01-29 05:07:04 +00:00
if ( f ) {
2009-06-16 18:54:30 +00:00
struct ast_frame * readq_tail = AST_LIST_LAST ( & chan - > readq ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
struct ast_control_read_action_payload * read_action_payload ;
struct ast_party_connected_line connected ;
2009-06-16 18:54:30 +00:00
2006-01-29 05:07:04 +00:00
/* if the channel driver returned more than one frame, stuff the excess
2009-06-16 18:54:30 +00:00
into the readq for the next ast_read call
2006-01-29 05:07:04 +00:00
*/
2006-08-29 20:50:36 +00:00
if ( AST_LIST_NEXT ( f , frame_list ) ) {
2009-06-16 18:54:30 +00:00
ast_queue_frame ( chan , AST_LIST_NEXT ( f , frame_list ) ) ;
ast_frfree ( AST_LIST_NEXT ( f , frame_list ) ) ;
2006-08-29 20:50:36 +00:00
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 :
2009-11-04 14:05:12 +00:00
if ( f - > subclass . integer = = 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 ;
2010-09-14 18:04:21 +00:00
} else if ( prestate = = AST_STATE_UP & & ast_bridged_channel ( chan ) ) {
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. */
2009-06-26 15:28:53 +00:00
ast_cel_report_event ( chan , AST_CEL_ANSWER , NULL , NULL , NULL ) ;
2006-01-29 05:07:04 +00:00
}
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
} else if ( f - > subclass . integer = = AST_CONTROL_READ_ACTION ) {
read_action_payload = f - > data . ptr ;
switch ( read_action_payload - > action ) {
case AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO :
2010-05-25 16:23:51 +00:00
ast_party_connected_line_init ( & connected ) ;
ast_party_connected_line_copy ( & connected , & chan - > connected ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
if ( ast_connected_line_parse_data ( read_action_payload - > payload ,
2010-05-25 16:23:51 +00:00
read_action_payload - > payload_size , & connected ) ) {
ast_party_connected_line_free ( & connected ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
break ;
}
if ( ast_channel_connected_line_macro ( NULL , chan , & connected , 1 , 0 ) ) {
ast_indicate_data ( chan , AST_CONTROL_CONNECTED_LINE ,
2010-05-25 16:23:51 +00:00
read_action_payload - > payload ,
read_action_payload - > payload_size ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
}
2010-05-25 16:23:51 +00:00
ast_party_connected_line_free ( & connected ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
break ;
}
ast_frfree ( f ) ;
f = & ast_null_frame ;
2006-01-29 05:07:04 +00:00
}
break ;
2006-08-31 01:59:02 +00:00
case AST_FRAME_DTMF_END :
2009-11-04 14:05:12 +00:00
send_dtmf_event ( chan , " Received " , f - > subclass . integer , " No " , " Yes " ) ;
ast_log ( LOG_DTMF , " DTMF end '%c' received on %s, duration %ld ms \n " , f - > subclass . integer , chan - > name , f - > len ) ;
2008-12-12 13:55:30 +00:00
/* Queue it up if DTMF is deferred, or if DTMF emulation is forced. */
if ( ast_test_flag ( chan , AST_FLAG_DEFER_DTMF ) | | ast_test_flag ( chan , AST_FLAG_EMULATE_DTMF ) ) {
queue_dtmf_readq ( chan , f ) ;
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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 */
2008-12-12 13:55:30 +00:00
queue_dtmf_readq ( chan , f ) ;
2007-04-24 19:03:16 +00:00
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 ) ;
2009-11-04 14:05:12 +00:00
chan - > emulate_dtmf_digit = f - > subclass . integer ;
2007-04-24 19:03:16 +00:00
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: https://origsvn.digium.com/svn/asterisk/trunk@62943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 ;
2009-11-04 14:05:12 +00:00
ast_log ( LOG_DTMF , " DTMF begin emulation of '%c' with duration %u queued on %s \n " , f - > subclass . integer , 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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 ) ) {
2009-11-04 14:05:12 +00:00
ast_log ( LOG_DTMF , " DTMF end accepted with begin '%c' on %s \n " , f - > subclass . integer , 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 ) ;
2010-07-16 17:30:39 +00:00
/* detect tones that were received on
* the wire with durations shorter than
* AST_MIN_DTMF_DURATION and set f - > len
* to the actual duration of the DTMF
* frames on the wire . This will cause
* dtmf emulation to be triggered later
* on .
*/
if ( ast_tvdiff_ms ( now , chan - > dtmf_tv ) < AST_MIN_DTMF_DURATION ) {
f - > len = ast_tvdiff_ms ( now , chan - > dtmf_tv ) ;
2010-07-16 18:05:01 +00:00
ast_log ( LOG_DTMF , " DTMF end '%c' detected to have actual duration %ld on the wire, emulation will be triggered on %s \n " , f - > subclass . integer , f - > len , chan - > name ) ;
2010-07-16 17:30:39 +00:00
}
2007-08-06 21:43:09 +00:00
} else if ( ! f - > len ) {
2009-11-04 14:05:12 +00:00
ast_log ( LOG_DTMF , " DTMF end accepted without begin '%c' on %s \n " , f - > subclass . integer , 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 ) ) {
2009-11-04 14:05:12 +00:00
ast_log ( LOG_DTMF , " DTMF end '%c' has duration %ld but want minimum %d, emulating on %s \n " , f - > subclass . integer , f - > len , AST_MIN_DTMF_DURATION , chan - > name ) ;
2007-04-23 17:58:15 +00:00
ast_set_flag ( chan , AST_FLAG_EMULATE_DTMF ) ;
2009-11-04 14:05:12 +00:00
chan - > emulate_dtmf_digit = f - > subclass . integer ;
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 {
2009-11-04 14:05:12 +00:00
ast_log ( LOG_DTMF , " DTMF end passthrough '%c' on %s \n " , f - > subclass . integer , 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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19 18:06:03 +00:00
}
2006-01-29 05:07:04 +00:00
break ;
case AST_FRAME_DTMF_BEGIN :
2009-11-04 14:05:12 +00:00
send_dtmf_event ( chan , " Received " , f - > subclass . integer , " Yes " , " No " ) ;
ast_log ( LOG_DTMF , " DTMF begin '%c' received on %s \n " , f - > subclass . integer , 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 ) ) {
2009-11-04 14:05:12 +00:00
ast_log ( LOG_DTMF , " DTMF begin ignored '%c' on %s \n " , f - > subclass . integer , 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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 ( ) ;
2009-11-04 14:05:12 +00:00
ast_log ( LOG_DTMF , " DTMF begin passthrough '%c' on %s \n " , f - > subclass . integer , 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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@62943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@62943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@62943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@62943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-03 15:23:44 +00:00
f - > frametype = AST_FRAME_DTMF_END ;
2009-11-04 14:05:12 +00:00
f - > subclass . integer = chan - > emulate_dtmf_digit ;
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: https://origsvn.digium.com/svn/asterisk/trunk@62943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-03 15:23:44 +00:00
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 ;
2009-11-04 14:05:12 +00:00
ast_log ( LOG_DTMF , " DTMF end emulation of '%c' queued on %s \n " , f - > subclass . integer , chan - > name ) ;
2009-04-01 13:48:26 +00:00
if ( chan - > audiohooks ) {
struct ast_frame * old_frame = f ;
f = ast_audiohook_write_list ( chan , chan - > audiohooks , AST_AUDIOHOOK_DIRECTION_READ , f ) ;
if ( old_frame ! = f ) {
ast_frfree ( old_frame ) ;
}
}
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: https://origsvn.digium.com/svn/asterisk/trunk@62943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@62943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@62943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 ;
2009-11-04 14:05:12 +00:00
f - > subclass . integer = 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 ) ;
}
2009-11-04 14:05:12 +00:00
ast_log ( LOG_DTMF , " DTMF end emulation of '%c' queued on %s \n " , f - > subclass . integer , 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: https://origsvn.digium.com/svn/asterisk/trunk@62943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 ;
}
2009-11-04 14:05:12 +00:00
} else if ( ( f - > frametype = = AST_FRAME_VOICE ) & & ! ( f - > subclass . codec & 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 " ,
2009-11-04 14:05:12 +00:00
chan - > name , ast_getformatname ( f - > subclass . codec ) , 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 ) {
2010-02-16 17:07:41 +00:00
jump = calc_monitor_jump ( ( chan - > outsmpl - chan - > insmpl ) , ast_format_rate ( f - > subclass . codec ) , ast_format_rate ( chan - > monitor - > read_stream - > fmt - > format ) ) ;
2007-01-24 01:00:57 +00:00
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 " ) ;
2010-02-16 17:07:41 +00:00
chan - > insmpl + = ( chan - > outsmpl - chan - > insmpl ) + f - > samples ;
2006-01-29 05:07:04 +00:00
} else
chan - > insmpl + = f - > samples ;
2003-08-28 20:02:10 +00:00
# else
2010-02-16 17:07:41 +00:00
int jump = calc_monitor_jump ( ( chan - > outsmpl - chan - > insmpl ) , ast_format_rate ( f - > subclass . codec ) , ast_format_rate ( chan - > monitor - > read_stream - > fmt - > format ) ) ;
2006-01-29 05:07:04 +00:00
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 " ) ;
2010-02-16 17:07:41 +00:00
chan - > insmpl + = chan - > outsmpl - chan - > insmpl ;
2006-01-29 05:07:04 +00:00
} 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 " ) ;
}
}
2009-06-16 18:54:30 +00:00
if ( chan - > readtrans & & ( f = ast_translate ( chan - > readtrans , f , 1 ) ) = = NULL ) {
2006-04-16 13:58:43 +00:00
f = & ast_null_frame ;
2009-06-16 18:54:30 +00:00
}
/* it is possible for the translation process on chan->readtrans to have
produced multiple frames from the single input frame we passed it ; if
this happens , queue the additional frames * before * the frames we may
have queued earlier . if the readq was empty , put them at the head of
the queue , and if it was not , put them just after the frame that was
at the end of the queue .
*/
if ( AST_LIST_NEXT ( f , frame_list ) ) {
if ( ! readq_tail ) {
ast_queue_frame_head ( chan , AST_LIST_NEXT ( f , frame_list ) ) ;
} else {
__ast_queue_frame ( chan , AST_LIST_NEXT ( f , frame_list ) , 0 , readq_tail ) ;
}
ast_frfree ( AST_LIST_NEXT ( f , frame_list ) ) ;
AST_LIST_NEXT ( f , frame_list ) = NULL ;
}
/* 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
}
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
break ;
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 */
2010-11-22 19:36:10 +00:00
if ( ! chan - > _softhangup ) {
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 )
2009-11-04 14:05:12 +00:00
chan - > generator - > digit ( chan , f - > subclass . integer ) ;
2007-05-22 18:52:59 +00:00
2010-07-27 20:57:00 +00:00
if ( chan - > audiohooks & & ast_audiohook_write_list_empty ( chan - > audiohooks ) ) {
/* The list gets recreated if audiohooks are added again later */
ast_audiohook_detach_list ( chan - > audiohooks ) ;
chan - > audiohooks = NULL ;
}
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 )
{
2010-04-02 23:30:58 +00:00
return ( ast_opt_internal_timing & & chan - > timingfd > - 1 ) ;
2006-03-30 06:07:04 +00:00
}
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 ) ;
}
2008-12-15 14:40:24 +00:00
static int attribute_const is_visible_indication ( enum ast_control_frame_type condition )
2001-05-07 00:43:32 +00:00
{
2008-12-15 14:40:24 +00:00
/* Don't include a default case here so that we get compiler warnings
* when a new type is added . */
switch ( condition ) {
case AST_CONTROL_PROGRESS :
case AST_CONTROL_PROCEEDING :
case AST_CONTROL_VIDUPDATE :
case AST_CONTROL_SRCUPDATE :
2010-03-12 22:04:51 +00:00
case AST_CONTROL_SRCCHANGE :
2008-12-15 14:40:24 +00:00
case AST_CONTROL_RADIO_KEY :
case AST_CONTROL_RADIO_UNKEY :
case AST_CONTROL_OPTION :
case AST_CONTROL_WINK :
case AST_CONTROL_FLASH :
case AST_CONTROL_OFFHOOK :
case AST_CONTROL_TAKEOFFHOOK :
case AST_CONTROL_ANSWER :
case AST_CONTROL_HANGUP :
2009-04-03 22:41:46 +00:00
case AST_CONTROL_CONNECTED_LINE :
case AST_CONTROL_REDIRECTING :
2009-04-03 16:47:27 +00:00
case AST_CONTROL_TRANSFER :
2009-06-26 19:27:24 +00:00
case AST_CONTROL_T38_PARAMETERS :
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-23 21:57:24 +00:00
case _XXX_AST_CONTROL_T38 :
Merge Call completion support into trunk.
From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:
1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
multiple calls to the same device. This proved to not be such a good idea
when implementing protocol-specific monitors, and so we ended up using one
monitor per-device per-call.
3. There are some configuration options which were conceived after the document
was written. These are documented in the ccss.conf.sample that is on this
review request.
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.
This implements CCBS and CCNR in several flavors.
First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.
Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:
* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
what is defined in the referenced draft.
* Implementation of the draft required support for SIP PUBLISH. I attempted to write
this in a generic-enough fashion such that if someone were to want to write PUBLISH
support for other event packages, such as dialog-state or presence, most of the effort
would be in writing callbacks specific to the event package.
* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
parser. The PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly. The rest of the
PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
code. In other words, while there is PIDF support here, it is not in any state
where it could easily be applied to other event packages as is.
Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.
Review: https://reviewboard.asterisk.org/r/523
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09 15:31:32 +00:00
case AST_CONTROL_CC :
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
case AST_CONTROL_READ_ACTION :
Generic Advice of Charge.
Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages
Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup
AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.
SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option.
IAX AOC Support
- Natively supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type
DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
example usage:
;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))
Review: https://reviewboard.asterisk.org/r/552/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02 18:10:15 +00:00
case AST_CONTROL_AOC :
2010-11-22 19:36:10 +00:00
case AST_CONTROL_END_OF_Q :
2011-09-16 19:53:40 +00:00
case AST_CONTROL_UPDATE_RTP_PEER :
2009-04-03 22:41:46 +00:00
break ;
2008-12-15 14:40:24 +00:00
2011-09-09 16:09:09 +00:00
case AST_CONTROL_INCOMPLETE :
2008-12-15 14:40:24 +00:00
case AST_CONTROL_CONGESTION :
case AST_CONTROL_BUSY :
case AST_CONTROL_RINGING :
case AST_CONTROL_RING :
case AST_CONTROL_HOLD :
2010-10-22 15:47:08 +00:00
/* You can hear these */
2008-12-15 14:40:24 +00:00
return 1 ;
2010-10-22 15:47:08 +00:00
case AST_CONTROL_UNHOLD :
/* This is a special case. You stop hearing this. */
break ;
2008-12-15 14:40:24 +00:00
}
return 0 ;
}
int ast_indicate_data ( struct ast_channel * chan , int _condition ,
const void * data , size_t datalen )
{
/* By using an enum, we'll get compiler warnings for values not handled
* in switch statements . */
enum ast_control_frame_type condition = _condition ;
2009-02-17 20:41:24 +00:00
struct ast_tone_zone_sound * ts = NULL ;
2009-04-03 22:41:46 +00:00
int res ;
2010-09-20 22:09:16 +00:00
/* this frame is used by framehooks. if it is set, we must free it at the end of this function */
struct ast_frame * awesome_frame = NULL ;
2005-07-25 22:13:32 +00:00
2006-04-19 06:58:20 +00:00
ast_channel_lock ( chan ) ;
2008-12-15 14:40:24 +00:00
/* Don't bother if the channel is about to go away, anyway. */
2006-01-24 22:38:17 +00:00
if ( ast_test_flag ( chan , AST_FLAG_ZOMBIE ) | | ast_check_hangup ( chan ) ) {
2010-09-20 22:09:16 +00:00
res = - 1 ;
goto indicate_cleanup ;
}
if ( ! ast_framehook_list_is_empty ( chan - > framehooks ) ) {
/* Do framehooks now, do it, go, go now */
struct ast_frame frame = {
. frametype = AST_FRAME_CONTROL ,
. subclass . integer = condition ,
. data . ptr = ( void * ) data , /* this cast from const is only okay because we do the ast_frdup below */
. datalen = datalen
} ;
/* we have now committed to freeing this frame */
awesome_frame = ast_frdup ( & frame ) ;
/* who knows what we will get back! the anticipation is killing me. */
2011-05-09 14:09:38 +00:00
if ( ! ( awesome_frame = ast_framehook_list_write_event ( chan - > framehooks , awesome_frame ) )
| | awesome_frame - > frametype ! = AST_FRAME_CONTROL ) {
2010-09-20 22:09:16 +00:00
res = 0 ;
goto indicate_cleanup ;
}
condition = awesome_frame - > subclass . integer ;
data = awesome_frame - > data . ptr ;
datalen = awesome_frame - > datalen ;
2006-01-24 22:38:17 +00:00
}
2010-09-20 22:09:16 +00:00
2009-04-03 22:41:46 +00:00
switch ( condition ) {
case AST_CONTROL_CONNECTED_LINE :
{
struct ast_party_connected_line connected ;
ast_party_connected_line_set_init ( & connected , & chan - > connected ) ;
res = ast_connected_line_parse_data ( data , datalen , & connected ) ;
if ( ! res ) {
2010-07-14 15:48:36 +00:00
ast_channel_set_connected_line ( chan , & connected , NULL ) ;
2009-04-03 22:41:46 +00:00
}
ast_party_connected_line_free ( & connected ) ;
}
break ;
case AST_CONTROL_REDIRECTING :
{
struct ast_party_redirecting redirecting ;
2008-12-15 14:40:24 +00:00
2009-04-03 22:41:46 +00:00
ast_party_redirecting_set_init ( & redirecting , & chan - > redirecting ) ;
res = ast_redirecting_parse_data ( data , datalen , & redirecting ) ;
if ( ! res ) {
2010-07-14 15:48:36 +00:00
ast_channel_set_redirecting ( chan , & redirecting , NULL ) ;
2009-04-03 22:41:46 +00:00
}
ast_party_redirecting_free ( & redirecting ) ;
}
break ;
default :
break ;
}
2009-07-06 13:38:29 +00:00
2010-10-22 15:47:08 +00:00
if ( is_visible_indication ( condition ) ) {
/* A new visible indication is requested. */
chan - > visible_indication = condition ;
} else if ( condition = = AST_CONTROL_UNHOLD | | _condition < 0 ) {
/* Visible indication is cleared/stopped. */
chan - > visible_indication = 0 ;
}
2008-12-15 14:40:24 +00:00
if ( chan - > tech - > indicate ) {
/* See if the channel driver can handle this condition. */
2006-05-10 12:24:11 +00:00
res = chan - > tech - > indicate ( chan , condition , data , datalen ) ;
2009-04-03 22:41:46 +00:00
} else {
res = - 1 ;
2008-12-15 14:40:24 +00:00
}
2009-07-06 13:38:29 +00:00
if ( ! res ) {
2008-12-15 14:40:24 +00:00
/* The channel driver successfully handled this indication */
2010-09-20 22:09:16 +00:00
res = 0 ;
goto indicate_cleanup ;
2008-12-15 14:40:24 +00:00
}
/* The channel driver does not support this indication, let's fake
* it by doing our own tone generation if applicable . */
2008-12-23 04:32:15 +00:00
/*!\note If we compare the enumeration type, which does not have any
* negative constants , the compiler may optimize this code away .
* Therefore , we must perform an integer comparison here . */
if ( _condition < 0 ) {
2008-12-15 14:40:24 +00:00
/* Stop any tones that are playing */
ast_playtones_stop ( chan ) ;
2010-09-20 22:09:16 +00:00
res = 0 ;
goto indicate_cleanup ;
2008-12-15 14:40:24 +00:00
}
/* Handle conditions that we have tones for. */
switch ( condition ) {
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-23 21:57:24 +00:00
case _XXX_AST_CONTROL_T38 :
/* deprecated T.38 control frame */
2010-09-20 22:09:16 +00:00
res = - 1 ;
goto indicate_cleanup ;
2009-07-06 13:38:29 +00:00
case AST_CONTROL_T38_PARAMETERS :
/* there is no way to provide 'default' behavior for these
* control frames , so we need to return failure , but there
* is also no value in the log message below being emitted
* since failure to handle these frames is not an ' error '
Improve handling of T.38 re-INVITEs that arrive before a T.38-capable
application is executing on a channel.
This patch addresses an issue found during working with end-users
using res_fax. If an incoming call is answered in the dialplan, or
jumps to the 'fax' extension due to reception of a CNG tone (with
faxdetect enabled), and then the remote endpoint sends a T.38
re-INVITE, it is possible for the channel's T.38 state to be
'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately,
even if the application wants to use T.38, it can't respond to the
peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS
control frame that chan_sip sent originally has been lost, and the
application needs the content of that frame to be able to formulate a
reply.
This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS,
AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip
will re-send the original control frame (with
AST_T38_REQUEST_NEGOTIATE as the request type), and the application
can respond as normal. If this occurs within the five second timeout
in chan_sip, the automatic cancellation of the peer reinvite will be
stopped, and the application will 'own' the negotiation process from
that point onwards.
This also improves the code path in chan_sip to allow sip_indicate(),
when called for AST_CONTROL_T38_PARAMETERS, to be able to return a
non-zero response, which should have been in place before since the
control frame *can* fail to be processed properly. It also modifies
ast_indicate() to return whatever result the channel driver returned
for this control frame, rather than converting all non-zero results
into '-1'. Finally, the new request type intentionally returns a
positive value, so that an application that sends
AST_T38_REQUEST_PARMS can know for certain whether the channel driver
accepted it and will be replying with a control frame of its own, or
whether it was ignored (if the sip_indicate()/ast_indicate() path had
properly supported failure responses before, this would not be
necessary).
This patch also modifies res_fax to take advantage of the new request.
In addition, this patch makes sip_t38_abort() actually lock the
private structure before doing its work... bad programmer, no donut.
This patch also enhances chan_sip's 'faxdetect' support to allow
triggering on T.38 re-INVITEs received as well as CNG tone detection.
Review: https://reviewboard.asterisk.org/r/556/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25 15:27:31 +00:00
* so just return right now . in addition , we want to return
* whatever value the channel driver returned , in case it
* has some meaning . */
2010-09-20 22:09:16 +00:00
goto indicate_cleanup ;
2008-12-15 14:40:24 +00:00
case AST_CONTROL_RINGING :
ts = ast_get_indication_tone ( chan - > zone , " ring " ) ;
2009-06-12 19:07:51 +00:00
/* It is common practice for channel drivers to return -1 if trying
* to indicate ringing on a channel which is up . The idea is to let the
* core generate the ringing inband . However , we don ' t want the
* warning message about not being able to handle the specific indication
* to print nor do we want ast_indicate_data to return an " error " for this
* condition
*/
if ( chan - > _state = = AST_STATE_UP ) {
res = 0 ;
}
2008-12-15 14:40:24 +00:00
break ;
case AST_CONTROL_BUSY :
ts = ast_get_indication_tone ( chan - > zone , " busy " ) ;
break ;
2011-09-09 16:09:09 +00:00
case AST_CONTROL_INCOMPLETE :
2008-12-15 14:40:24 +00:00
case AST_CONTROL_CONGESTION :
ts = ast_get_indication_tone ( chan - > zone , " congestion " ) ;
break ;
case AST_CONTROL_PROGRESS :
case AST_CONTROL_PROCEEDING :
case AST_CONTROL_VIDUPDATE :
case AST_CONTROL_SRCUPDATE :
2010-03-12 22:04:51 +00:00
case AST_CONTROL_SRCCHANGE :
2008-12-15 14:40:24 +00:00
case AST_CONTROL_RADIO_KEY :
case AST_CONTROL_RADIO_UNKEY :
case AST_CONTROL_OPTION :
case AST_CONTROL_WINK :
case AST_CONTROL_FLASH :
case AST_CONTROL_OFFHOOK :
case AST_CONTROL_TAKEOFFHOOK :
case AST_CONTROL_ANSWER :
case AST_CONTROL_HANGUP :
case AST_CONTROL_RING :
case AST_CONTROL_HOLD :
case AST_CONTROL_UNHOLD :
2009-04-03 16:47:27 +00:00
case AST_CONTROL_TRANSFER :
2009-04-03 22:41:46 +00:00
case AST_CONTROL_CONNECTED_LINE :
case AST_CONTROL_REDIRECTING :
Merge Call completion support into trunk.
From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:
1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
multiple calls to the same device. This proved to not be such a good idea
when implementing protocol-specific monitors, and so we ended up using one
monitor per-device per-call.
3. There are some configuration options which were conceived after the document
was written. These are documented in the ccss.conf.sample that is on this
review request.
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.
This implements CCBS and CCNR in several flavors.
First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.
Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:
* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
what is defined in the referenced draft.
* Implementation of the draft required support for SIP PUBLISH. I attempted to write
this in a generic-enough fashion such that if someone were to want to write PUBLISH
support for other event packages, such as dialog-state or presence, most of the effort
would be in writing callbacks specific to the event package.
* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
parser. The PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly. The rest of the
PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
code. In other words, while there is PIDF support here, it is not in any state
where it could easily be applied to other event packages as is.
Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.
Review: https://reviewboard.asterisk.org/r/523
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09 15:31:32 +00:00
case AST_CONTROL_CC :
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
case AST_CONTROL_READ_ACTION :
Generic Advice of Charge.
Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages
Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup
AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.
SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option.
IAX AOC Support
- Natively supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type
DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
example usage:
;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))
Review: https://reviewboard.asterisk.org/r/552/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02 18:10:15 +00:00
case AST_CONTROL_AOC :
2010-11-22 19:36:10 +00:00
case AST_CONTROL_END_OF_Q :
2011-09-16 19:53:40 +00:00
case AST_CONTROL_UPDATE_RTP_PEER :
2008-12-15 14:40:24 +00:00
/* Nothing left to do for these. */
res = 0 ;
break ;
}
2009-02-17 20:41:24 +00:00
if ( ts ) {
2008-12-15 14:40:24 +00:00
/* We have a tone to play, yay. */
ast_debug ( 1 , " Driver for channel '%s' does not support indication %d, emulating it \n " , chan - > name , condition ) ;
Merged revisions 296001 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines
Merged revisions 296000 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines
Handle failures building translation paths more effectively.
The problem scenario occurred on a heavily loaded system that was using the
codec_dahdi module and exceeded the hardware transcoding capacity. The failure
mode at that point was not good. The report came in to us as an Asterisk
lock-up. The "core show locks" shows a ton of threads locked up (but no
obvious deadlock). Upon deeper investigation, when the system is in this
state, the CPU was maxed out. The CPU was being consumed by the Asterisk
logger spewing messages on every audio frame for calls set up after transcoder
capacity was reached.
The purpose of this patch is to make Asterisk handle failures to create a
translation path in a more graceful manner. If we can't translate, then the
call just needs to be dropped, as it's not going to work. These are the
changes:
1) In set_format() of channel.c (which is called by set_read_format() and
set_write_format()), it was ignoring if ast_translator_build_path() failed and
returned NULL. It now pays attention to that case and returns a result
reflecting failure. With this change in place, the bridging code will
immediately detect a failure and end the bridge instead of proceeding to try to
bridge frames that can't be translated and making channel drivers freak out by
sending them frames in a format they weren't expecting.
2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was
ignored. It is now reflected in the return value of the function. This didn't
turn out to have any affect on the bug, but seemed like a good change to leave
in.
3) In app_dial(), when only sending a call to a single endpoint, it will
attempt to do some bridging of its own of early audio. It uses
make_compatible() when it's going to do this. However, it ignored failure from
make compatible. So, even with the fix from #1, if there was early audio going
through app_dial, there would still be a period of invalid frames passing
through. After detecting failure here, Dial() exits.
ABE-2658
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296002 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24 17:13:08 +00:00
res = ast_playtones_start ( chan , 0 , ts - > data , 1 ) ;
2009-02-17 20:41:24 +00:00
ts = ast_tone_zone_sound_unref ( ts ) ;
2008-12-15 14:40:24 +00:00
}
if ( res ) {
/* not handled */
ast_log ( LOG_WARNING , " Unable to handle indication %d for '%s' \n " , condition , chan - > name ) ;
}
2007-12-03 18:44:16 +00:00
2010-09-20 22:09:16 +00:00
indicate_cleanup :
2010-10-22 15:47:08 +00:00
ast_channel_unlock ( chan ) ;
2010-09-20 22:09:16 +00:00
if ( awesome_frame ) {
ast_frfree ( awesome_frame ) ;
}
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 */
2009-11-04 14:05:12 +00:00
if ( f - > frametype = = AST_FRAME_CONTROL & & f - > subclass . integer = = AST_CONTROL_HANGUP )
2005-06-23 21:28:09 +00:00
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 ;
2011-02-03 00:24:40 +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 */
2011-02-03 00:24:40 +00:00
if ( ast_test_flag ( chan , AST_FLAG_ZOMBIE ) | | ast_check_hangup ( chan ) ) {
ast_channel_unlock ( chan ) ;
2001-05-07 00:43:32 +00:00
return - 1 ;
2011-02-03 00:24:40 +00:00
}
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 ) ;
2011-02-03 00:24:40 +00:00
ast_channel_unlock ( chan ) ;
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 . */
2009-05-21 21:13:09 +00:00
static const char * const dtmf_tones [ ] = {
2007-05-09 19:21:35 +00:00
" 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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 ) ;
2009-11-04 14:05:12 +00:00
a . subclass . codec = chan - > rawwriteformat ;
2008-05-22 16:29:54 +00:00
a . data . ptr = nothing + AST_FRIENDLY_OFFSET ;
2010-07-12 17:21:01 +00:00
a . src = " ast_prod " ; /* this better match check in ast_write */
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 ;
}
Fix transcode_via_sln option with SIP calls and improve PLC usage.
From reviewboard:
The problem here is a bit complex, so try to bear with me...
It was noticed by a Digium customer that generic PLC (as configured in
codecs.conf) did not appear to actually be having any sort of benefit when
packet loss was introduced on an RTP stream. I reproduced this issue myself
by streaming a file across an RTP stream and dropping approx. 5% of the
RTP packets. I saw no real difference between when PLC was enabled or disabled
when using wireshark to analyze the RTP streams.
After analyzing what was going on, it became clear that one of the problems
faced was that when running my tests, the translation paths were being set
up in such a way that PLC could not possibly work as expected. To illustrate,
if packets are lost on channel A's read stream, then we expect that PLC will
be applied to channel B's write stream. The problem is that generic PLC can
only be done when there is a translation path that moves from some codec to
SLINEAR. When I would run my tests, I found that every single time, read
and write translation paths would be set up on channel A instead of channel
B. There appeared to be no real way to predict which channel the translation
paths would be set up on.
This is where Kevin swooped in to let me know about the transcode_via_sln
option in asterisk.conf. It is supposed to work by placing a read translation
path on both channels from the channel's rawreadformat to SLINEAR. It also
will place a write translation path on both channels from SLINEAR to the
channel's rawwriteformat. Using this option allows one to predictably set up
translation paths on all channels. There are two problems with this, though.
First and foremost, the transcode_via_sln option did not appear to be working
properly when I was placing a SIP call between two endpoints which did not
share any common formats. Second, even if this option were to work, for PLC
to be applied, there had to be a write translation path that would go from
some format to SLINEAR. It would not work properly if the starting format
of translation was SLINEAR.
The one-line change presented in this review request in chan_sip.c fixed the
first issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the format passed to
sip_request_call. This is nativeformats of the inbound channel. Because of this,
when ast_channel_make_compatible was called by app_dial, both channels already
had compatibly read and write formats. Thus, no translation path was set up at
the time. My change is to set the jointcapability of the sip_pvt created during
sip_request_call to the intersection of the inbound channel's nativeformats and
the configured peer capability that we determined during the earlier call to
create_addr. Doing this got the translation paths set up as expected when using
transcode_via_sln.
The changes presented in channel.c fixed the second issue for me. First and
foremost, when Asterisk is started, we'll read codecs.conf to see the value of
the genericplc option. If this option is set, and ast_write is called for a
frame with no data, then we will attempt to fill in the missing samples for
the frame. The implementation uses a channel datastore for maintaining the
PLC state and for creating a buffer to store PLC samples in. Even when we
receive a frame with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a basis for when
it comes time to actually do a PLC fill-in.
So, reviewers, now I ask for your help. First off, there's the one line change
in chan_sip that I have put in. Is it right? By my logic it seems correct, but
I'm sure someone can tell me why it is not going to work. This is probably the
change I'm least concerned about, though. What concerns me much more is the
set of changes in channel.c. First off, am I even doing it right? When I run
tests, I can clearly see that when PLC is activated, I see a significant increase
in RTP traffic where I would expect it to be. However, in my humble opinion, the
audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to
me than when no PLC is used at all. I need someone to review the logic I have used
to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic
is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm
sure someone can point out somewhere where I've done something incorrectly.
As I was writing this review request up, I decided to give the code a test run under
valgrind, and I find that for some reason, calls to plc_rx are causing some invalid
reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around
a bit to see why that is the case. If it's obvious to someone reviewing, speak up!
Finally, I have one other proposal that is not reflected in my code review. Since
without transcode_via_sln set, one cannot predict or control where a translation
path will be up, it seems to me that the current practice of using PLC only when
transcoding to SLINEAR is not useful. I recommend that once it has been determined
that the method used in this code review is correct and works as expected, then
the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 21:29:08 +00:00
struct plc_ds {
/* A buffer in which to store SLIN PLC
* samples generated by the generic PLC
* functionality in plc . c
*/
int16_t * samples_buf ;
/* The current number of samples in the
* samples_buf
*/
size_t num_samples ;
plc_state_t plc_state ;
} ;
static void plc_ds_destroy ( void * data )
{
struct plc_ds * plc = data ;
ast_free ( plc - > samples_buf ) ;
ast_free ( plc ) ;
}
static struct ast_datastore_info plc_ds_info = {
. type = " plc " ,
. destroy = plc_ds_destroy ,
} ;
static void adjust_frame_for_plc ( struct ast_channel * chan , struct ast_frame * frame , struct ast_datastore * datastore )
{
int num_new_samples = frame - > samples ;
struct plc_ds * plc = datastore - > data ;
2010-06-10 19:34:03 +00:00
/* As a general note, let me explain the somewhat odd calculations used when taking
* the frame offset into account here . According to documentation in frame . h , the frame ' s
* offset field indicates the number of bytes that the audio is offset . The plc - > samples_buf
* is not an array of bytes , but rather an array of 16 - bit integers since it holds SLIN
* samples . So I had two choices to make here with the offset .
*
* 1. Make the offset AST_FRIENDLY_OFFSET bytes . The main downside for this is that
* I can ' t just add AST_FRIENDLY_OFFSET to the plc - > samples_buf and have the pointer
* arithmetic come out right . I would have to do some odd casting or division for this to
* work as I wanted .
* 2. Make the offset AST_FRIENDLY_OFFSET * 2 bytes . This allows the pointer arithmetic
* to work out better with the plc - > samples_buf . The downside here is that the buffer ' s
* allocation contains an extra 64 bytes of unused space .
*
* I decided to go with option 2. This is why in the calloc statement and the statement that
* sets the frame ' s offset , AST_FRIENDLY_OFFSET is multiplied by 2.
*/
2010-05-24 16:10:09 +00:00
2010-06-10 19:34:03 +00:00
/* If this audio frame has no samples to fill in, ignore it */
2010-05-24 16:10:09 +00:00
if ( ! num_new_samples ) {
return ;
}
Fix transcode_via_sln option with SIP calls and improve PLC usage.
From reviewboard:
The problem here is a bit complex, so try to bear with me...
It was noticed by a Digium customer that generic PLC (as configured in
codecs.conf) did not appear to actually be having any sort of benefit when
packet loss was introduced on an RTP stream. I reproduced this issue myself
by streaming a file across an RTP stream and dropping approx. 5% of the
RTP packets. I saw no real difference between when PLC was enabled or disabled
when using wireshark to analyze the RTP streams.
After analyzing what was going on, it became clear that one of the problems
faced was that when running my tests, the translation paths were being set
up in such a way that PLC could not possibly work as expected. To illustrate,
if packets are lost on channel A's read stream, then we expect that PLC will
be applied to channel B's write stream. The problem is that generic PLC can
only be done when there is a translation path that moves from some codec to
SLINEAR. When I would run my tests, I found that every single time, read
and write translation paths would be set up on channel A instead of channel
B. There appeared to be no real way to predict which channel the translation
paths would be set up on.
This is where Kevin swooped in to let me know about the transcode_via_sln
option in asterisk.conf. It is supposed to work by placing a read translation
path on both channels from the channel's rawreadformat to SLINEAR. It also
will place a write translation path on both channels from SLINEAR to the
channel's rawwriteformat. Using this option allows one to predictably set up
translation paths on all channels. There are two problems with this, though.
First and foremost, the transcode_via_sln option did not appear to be working
properly when I was placing a SIP call between two endpoints which did not
share any common formats. Second, even if this option were to work, for PLC
to be applied, there had to be a write translation path that would go from
some format to SLINEAR. It would not work properly if the starting format
of translation was SLINEAR.
The one-line change presented in this review request in chan_sip.c fixed the
first issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the format passed to
sip_request_call. This is nativeformats of the inbound channel. Because of this,
when ast_channel_make_compatible was called by app_dial, both channels already
had compatibly read and write formats. Thus, no translation path was set up at
the time. My change is to set the jointcapability of the sip_pvt created during
sip_request_call to the intersection of the inbound channel's nativeformats and
the configured peer capability that we determined during the earlier call to
create_addr. Doing this got the translation paths set up as expected when using
transcode_via_sln.
The changes presented in channel.c fixed the second issue for me. First and
foremost, when Asterisk is started, we'll read codecs.conf to see the value of
the genericplc option. If this option is set, and ast_write is called for a
frame with no data, then we will attempt to fill in the missing samples for
the frame. The implementation uses a channel datastore for maintaining the
PLC state and for creating a buffer to store PLC samples in. Even when we
receive a frame with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a basis for when
it comes time to actually do a PLC fill-in.
So, reviewers, now I ask for your help. First off, there's the one line change
in chan_sip that I have put in. Is it right? By my logic it seems correct, but
I'm sure someone can tell me why it is not going to work. This is probably the
change I'm least concerned about, though. What concerns me much more is the
set of changes in channel.c. First off, am I even doing it right? When I run
tests, I can clearly see that when PLC is activated, I see a significant increase
in RTP traffic where I would expect it to be. However, in my humble opinion, the
audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to
me than when no PLC is used at all. I need someone to review the logic I have used
to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic
is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm
sure someone can point out somewhere where I've done something incorrectly.
As I was writing this review request up, I decided to give the code a test run under
valgrind, and I find that for some reason, calls to plc_rx are causing some invalid
reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around
a bit to see why that is the case. If it's obvious to someone reviewing, speak up!
Finally, I have one other proposal that is not reflected in my code review. Since
without transcode_via_sln set, one cannot predict or control where a translation
path will be up, it seems to me that the current practice of using PLC only when
transcoding to SLINEAR is not useful. I recommend that once it has been determined
that the method used in this code review is correct and works as expected, then
the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 21:29:08 +00:00
/* First, we need to be sure that our buffer is large enough to accomodate
* the samples we need to fill in . This will likely only occur on the first
* frame we write .
*/
if ( plc - > num_samples < num_new_samples ) {
ast_free ( plc - > samples_buf ) ;
2010-06-10 19:34:03 +00:00
plc - > samples_buf = ast_calloc ( 1 , ( num_new_samples * sizeof ( * plc - > samples_buf ) ) + ( AST_FRIENDLY_OFFSET * 2 ) ) ;
Fix transcode_via_sln option with SIP calls and improve PLC usage.
From reviewboard:
The problem here is a bit complex, so try to bear with me...
It was noticed by a Digium customer that generic PLC (as configured in
codecs.conf) did not appear to actually be having any sort of benefit when
packet loss was introduced on an RTP stream. I reproduced this issue myself
by streaming a file across an RTP stream and dropping approx. 5% of the
RTP packets. I saw no real difference between when PLC was enabled or disabled
when using wireshark to analyze the RTP streams.
After analyzing what was going on, it became clear that one of the problems
faced was that when running my tests, the translation paths were being set
up in such a way that PLC could not possibly work as expected. To illustrate,
if packets are lost on channel A's read stream, then we expect that PLC will
be applied to channel B's write stream. The problem is that generic PLC can
only be done when there is a translation path that moves from some codec to
SLINEAR. When I would run my tests, I found that every single time, read
and write translation paths would be set up on channel A instead of channel
B. There appeared to be no real way to predict which channel the translation
paths would be set up on.
This is where Kevin swooped in to let me know about the transcode_via_sln
option in asterisk.conf. It is supposed to work by placing a read translation
path on both channels from the channel's rawreadformat to SLINEAR. It also
will place a write translation path on both channels from SLINEAR to the
channel's rawwriteformat. Using this option allows one to predictably set up
translation paths on all channels. There are two problems with this, though.
First and foremost, the transcode_via_sln option did not appear to be working
properly when I was placing a SIP call between two endpoints which did not
share any common formats. Second, even if this option were to work, for PLC
to be applied, there had to be a write translation path that would go from
some format to SLINEAR. It would not work properly if the starting format
of translation was SLINEAR.
The one-line change presented in this review request in chan_sip.c fixed the
first issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the format passed to
sip_request_call. This is nativeformats of the inbound channel. Because of this,
when ast_channel_make_compatible was called by app_dial, both channels already
had compatibly read and write formats. Thus, no translation path was set up at
the time. My change is to set the jointcapability of the sip_pvt created during
sip_request_call to the intersection of the inbound channel's nativeformats and
the configured peer capability that we determined during the earlier call to
create_addr. Doing this got the translation paths set up as expected when using
transcode_via_sln.
The changes presented in channel.c fixed the second issue for me. First and
foremost, when Asterisk is started, we'll read codecs.conf to see the value of
the genericplc option. If this option is set, and ast_write is called for a
frame with no data, then we will attempt to fill in the missing samples for
the frame. The implementation uses a channel datastore for maintaining the
PLC state and for creating a buffer to store PLC samples in. Even when we
receive a frame with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a basis for when
it comes time to actually do a PLC fill-in.
So, reviewers, now I ask for your help. First off, there's the one line change
in chan_sip that I have put in. Is it right? By my logic it seems correct, but
I'm sure someone can tell me why it is not going to work. This is probably the
change I'm least concerned about, though. What concerns me much more is the
set of changes in channel.c. First off, am I even doing it right? When I run
tests, I can clearly see that when PLC is activated, I see a significant increase
in RTP traffic where I would expect it to be. However, in my humble opinion, the
audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to
me than when no PLC is used at all. I need someone to review the logic I have used
to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic
is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm
sure someone can point out somewhere where I've done something incorrectly.
As I was writing this review request up, I decided to give the code a test run under
valgrind, and I find that for some reason, calls to plc_rx are causing some invalid
reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around
a bit to see why that is the case. If it's obvious to someone reviewing, speak up!
Finally, I have one other proposal that is not reflected in my code review. Since
without transcode_via_sln set, one cannot predict or control where a translation
path will be up, it seems to me that the current practice of using PLC only when
transcoding to SLINEAR is not useful. I recommend that once it has been determined
that the method used in this code review is correct and works as expected, then
the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 21:29:08 +00:00
if ( ! plc - > samples_buf ) {
ast_channel_datastore_remove ( chan , datastore ) ;
ast_datastore_free ( datastore ) ;
return ;
}
plc - > num_samples = num_new_samples ;
}
if ( frame - > datalen = = 0 ) {
2010-06-10 19:34:03 +00:00
plc_fillin ( & plc - > plc_state , plc - > samples_buf + AST_FRIENDLY_OFFSET , frame - > samples ) ;
frame - > data . ptr = plc - > samples_buf + AST_FRIENDLY_OFFSET ;
Fix transcode_via_sln option with SIP calls and improve PLC usage.
From reviewboard:
The problem here is a bit complex, so try to bear with me...
It was noticed by a Digium customer that generic PLC (as configured in
codecs.conf) did not appear to actually be having any sort of benefit when
packet loss was introduced on an RTP stream. I reproduced this issue myself
by streaming a file across an RTP stream and dropping approx. 5% of the
RTP packets. I saw no real difference between when PLC was enabled or disabled
when using wireshark to analyze the RTP streams.
After analyzing what was going on, it became clear that one of the problems
faced was that when running my tests, the translation paths were being set
up in such a way that PLC could not possibly work as expected. To illustrate,
if packets are lost on channel A's read stream, then we expect that PLC will
be applied to channel B's write stream. The problem is that generic PLC can
only be done when there is a translation path that moves from some codec to
SLINEAR. When I would run my tests, I found that every single time, read
and write translation paths would be set up on channel A instead of channel
B. There appeared to be no real way to predict which channel the translation
paths would be set up on.
This is where Kevin swooped in to let me know about the transcode_via_sln
option in asterisk.conf. It is supposed to work by placing a read translation
path on both channels from the channel's rawreadformat to SLINEAR. It also
will place a write translation path on both channels from SLINEAR to the
channel's rawwriteformat. Using this option allows one to predictably set up
translation paths on all channels. There are two problems with this, though.
First and foremost, the transcode_via_sln option did not appear to be working
properly when I was placing a SIP call between two endpoints which did not
share any common formats. Second, even if this option were to work, for PLC
to be applied, there had to be a write translation path that would go from
some format to SLINEAR. It would not work properly if the starting format
of translation was SLINEAR.
The one-line change presented in this review request in chan_sip.c fixed the
first issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the format passed to
sip_request_call. This is nativeformats of the inbound channel. Because of this,
when ast_channel_make_compatible was called by app_dial, both channels already
had compatibly read and write formats. Thus, no translation path was set up at
the time. My change is to set the jointcapability of the sip_pvt created during
sip_request_call to the intersection of the inbound channel's nativeformats and
the configured peer capability that we determined during the earlier call to
create_addr. Doing this got the translation paths set up as expected when using
transcode_via_sln.
The changes presented in channel.c fixed the second issue for me. First and
foremost, when Asterisk is started, we'll read codecs.conf to see the value of
the genericplc option. If this option is set, and ast_write is called for a
frame with no data, then we will attempt to fill in the missing samples for
the frame. The implementation uses a channel datastore for maintaining the
PLC state and for creating a buffer to store PLC samples in. Even when we
receive a frame with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a basis for when
it comes time to actually do a PLC fill-in.
So, reviewers, now I ask for your help. First off, there's the one line change
in chan_sip that I have put in. Is it right? By my logic it seems correct, but
I'm sure someone can tell me why it is not going to work. This is probably the
change I'm least concerned about, though. What concerns me much more is the
set of changes in channel.c. First off, am I even doing it right? When I run
tests, I can clearly see that when PLC is activated, I see a significant increase
in RTP traffic where I would expect it to be. However, in my humble opinion, the
audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to
me than when no PLC is used at all. I need someone to review the logic I have used
to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic
is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm
sure someone can point out somewhere where I've done something incorrectly.
As I was writing this review request up, I decided to give the code a test run under
valgrind, and I find that for some reason, calls to plc_rx are causing some invalid
reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around
a bit to see why that is the case. If it's obvious to someone reviewing, speak up!
Finally, I have one other proposal that is not reflected in my code review. Since
without transcode_via_sln set, one cannot predict or control where a translation
path will be up, it seems to me that the current practice of using PLC only when
transcoding to SLINEAR is not useful. I recommend that once it has been determined
that the method used in this code review is correct and works as expected, then
the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 21:29:08 +00:00
frame - > datalen = num_new_samples * 2 ;
2010-06-10 19:34:03 +00:00
frame - > offset = AST_FRIENDLY_OFFSET * 2 ;
Fix transcode_via_sln option with SIP calls and improve PLC usage.
From reviewboard:
The problem here is a bit complex, so try to bear with me...
It was noticed by a Digium customer that generic PLC (as configured in
codecs.conf) did not appear to actually be having any sort of benefit when
packet loss was introduced on an RTP stream. I reproduced this issue myself
by streaming a file across an RTP stream and dropping approx. 5% of the
RTP packets. I saw no real difference between when PLC was enabled or disabled
when using wireshark to analyze the RTP streams.
After analyzing what was going on, it became clear that one of the problems
faced was that when running my tests, the translation paths were being set
up in such a way that PLC could not possibly work as expected. To illustrate,
if packets are lost on channel A's read stream, then we expect that PLC will
be applied to channel B's write stream. The problem is that generic PLC can
only be done when there is a translation path that moves from some codec to
SLINEAR. When I would run my tests, I found that every single time, read
and write translation paths would be set up on channel A instead of channel
B. There appeared to be no real way to predict which channel the translation
paths would be set up on.
This is where Kevin swooped in to let me know about the transcode_via_sln
option in asterisk.conf. It is supposed to work by placing a read translation
path on both channels from the channel's rawreadformat to SLINEAR. It also
will place a write translation path on both channels from SLINEAR to the
channel's rawwriteformat. Using this option allows one to predictably set up
translation paths on all channels. There are two problems with this, though.
First and foremost, the transcode_via_sln option did not appear to be working
properly when I was placing a SIP call between two endpoints which did not
share any common formats. Second, even if this option were to work, for PLC
to be applied, there had to be a write translation path that would go from
some format to SLINEAR. It would not work properly if the starting format
of translation was SLINEAR.
The one-line change presented in this review request in chan_sip.c fixed the
first issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the format passed to
sip_request_call. This is nativeformats of the inbound channel. Because of this,
when ast_channel_make_compatible was called by app_dial, both channels already
had compatibly read and write formats. Thus, no translation path was set up at
the time. My change is to set the jointcapability of the sip_pvt created during
sip_request_call to the intersection of the inbound channel's nativeformats and
the configured peer capability that we determined during the earlier call to
create_addr. Doing this got the translation paths set up as expected when using
transcode_via_sln.
The changes presented in channel.c fixed the second issue for me. First and
foremost, when Asterisk is started, we'll read codecs.conf to see the value of
the genericplc option. If this option is set, and ast_write is called for a
frame with no data, then we will attempt to fill in the missing samples for
the frame. The implementation uses a channel datastore for maintaining the
PLC state and for creating a buffer to store PLC samples in. Even when we
receive a frame with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a basis for when
it comes time to actually do a PLC fill-in.
So, reviewers, now I ask for your help. First off, there's the one line change
in chan_sip that I have put in. Is it right? By my logic it seems correct, but
I'm sure someone can tell me why it is not going to work. This is probably the
change I'm least concerned about, though. What concerns me much more is the
set of changes in channel.c. First off, am I even doing it right? When I run
tests, I can clearly see that when PLC is activated, I see a significant increase
in RTP traffic where I would expect it to be. However, in my humble opinion, the
audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to
me than when no PLC is used at all. I need someone to review the logic I have used
to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic
is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm
sure someone can point out somewhere where I've done something incorrectly.
As I was writing this review request up, I decided to give the code a test run under
valgrind, and I find that for some reason, calls to plc_rx are causing some invalid
reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around
a bit to see why that is the case. If it's obvious to someone reviewing, speak up!
Finally, I have one other proposal that is not reflected in my code review. Since
without transcode_via_sln set, one cannot predict or control where a translation
path will be up, it seems to me that the current practice of using PLC only when
transcoding to SLINEAR is not useful. I recommend that once it has been determined
that the method used in this code review is correct and works as expected, then
the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 21:29:08 +00:00
} else {
plc_rx ( & plc - > plc_state , frame - > data . ptr , frame - > samples ) ;
}
}
static void apply_plc ( struct ast_channel * chan , struct ast_frame * frame )
{
struct ast_datastore * datastore ;
struct plc_ds * plc ;
datastore = ast_channel_datastore_find ( chan , & plc_ds_info , NULL ) ;
if ( datastore ) {
plc = datastore - > data ;
adjust_frame_for_plc ( chan , frame , datastore ) ;
return ;
}
datastore = ast_datastore_alloc ( & plc_ds_info , NULL ) ;
if ( ! datastore ) {
return ;
}
plc = ast_calloc ( 1 , sizeof ( * plc ) ) ;
if ( ! plc ) {
ast_datastore_free ( datastore ) ;
return ;
}
datastore - > data = plc ;
ast_channel_datastore_add ( chan , datastore ) ;
adjust_frame_for_plc ( chan , frame , datastore ) ;
}
1999-12-04 21:35:07 +00:00
int ast_write ( struct ast_channel * chan , struct ast_frame * fr )
{
int res = - 1 ;
2009-06-16 18:54:30 +00:00
struct ast_frame * f = 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 */
2009-10-07 22:58:38 +00:00
if ( chan - > masq ) {
ast_channel_unlock ( chan ) ;
if ( ast_do_masquerade ( chan ) ) {
ast_log ( LOG_WARNING , " Failed to perform masquerade \n " ) ;
return res ; /* no need to goto done: chan is already unlocked for masq */
}
ast_channel_lock ( chan ) ;
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
}
2010-09-20 22:09:16 +00:00
/* Perform the framehook write event here. After the frame enters the framehook list
* there is no telling what will happen , how awesome is that ! ! ! */
if ( ! ( fr = ast_framehook_list_write_event ( chan - > framehooks , fr ) ) ) {
res = 0 ;
goto done ;
}
2010-07-15 13:51:11 +00:00
if ( chan - > generatordata & & ( ! fr - > src | | strcasecmp ( fr - > src , " ast_prod " ) ) ) {
2010-07-12 17:21:01 +00:00
if ( ast_test_flag ( chan , AST_FLAG_WRITE_INT ) ) {
ast_deactivate_generator ( chan ) ;
} 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 ) ;
2009-11-04 14:05:12 +00:00
res = ast_senddigit_end ( chan , fr - > subclass . integer , fr - > len ) ;
2007-05-02 23:00:07 +00:00
ast_channel_lock ( chan ) ;
CHECK_BLOCKING ( chan ) ;
2009-11-04 14:05:12 +00:00
} else if ( fr - > frametype = = AST_FRAME_CONTROL & & fr - > subclass . integer = = AST_CONTROL_UNHOLD ) {
2007-05-14 17:25:25 +00:00
/* 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 :
2009-11-04 14:05:12 +00:00
chan - > tech - > indicate ( chan , fr - > subclass . integer , 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 :
2009-11-04 14:05:12 +00:00
chan - > tech - > indicate ( chan , fr - > subclass . integer , 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 ;
}
2009-11-04 14:05:12 +00:00
send_dtmf_event ( chan , " Sent " , fr - > subclass . integer , " Yes " , " No " ) ;
2006-08-31 01:59:02 +00:00
ast_clear_flag ( chan , AST_FLAG_BLOCKING ) ;
ast_channel_unlock ( chan ) ;
2009-11-04 14:05:12 +00:00
res = ast_senddigit_begin ( chan , fr - > subclass . integer ) ;
2006-08-31 01:59:02 +00:00
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 ) {
2009-06-16 18:54:30 +00:00
struct ast_frame * new_frame = fr ;
new_frame = ast_audiohook_write_list ( chan , chan - > audiohooks , AST_AUDIOHOOK_DIRECTION_WRITE , fr ) ;
if ( new_frame ! = fr ) {
ast_frfree ( new_frame ) ;
}
2007-09-06 22:34:44 +00:00
}
2009-11-04 14:05:12 +00:00
send_dtmf_event ( chan , " Sent " , fr - > subclass . integer , " 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 ) ;
2009-11-04 14:05:12 +00:00
res = ast_senddigit_end ( chan , fr - > subclass . integer , 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 :
2009-11-04 14:05:12 +00:00
if ( fr - > subclass . integer = = AST_FORMAT_T140 ) {
2007-02-16 13:35:44 +00:00
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 :
2009-11-04 14:05:12 +00:00
chan - > tech - > send_html ( chan , fr - > subclass . integer , ( 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
Fix transcode_via_sln option with SIP calls and improve PLC usage.
From reviewboard:
The problem here is a bit complex, so try to bear with me...
It was noticed by a Digium customer that generic PLC (as configured in
codecs.conf) did not appear to actually be having any sort of benefit when
packet loss was introduced on an RTP stream. I reproduced this issue myself
by streaming a file across an RTP stream and dropping approx. 5% of the
RTP packets. I saw no real difference between when PLC was enabled or disabled
when using wireshark to analyze the RTP streams.
After analyzing what was going on, it became clear that one of the problems
faced was that when running my tests, the translation paths were being set
up in such a way that PLC could not possibly work as expected. To illustrate,
if packets are lost on channel A's read stream, then we expect that PLC will
be applied to channel B's write stream. The problem is that generic PLC can
only be done when there is a translation path that moves from some codec to
SLINEAR. When I would run my tests, I found that every single time, read
and write translation paths would be set up on channel A instead of channel
B. There appeared to be no real way to predict which channel the translation
paths would be set up on.
This is where Kevin swooped in to let me know about the transcode_via_sln
option in asterisk.conf. It is supposed to work by placing a read translation
path on both channels from the channel's rawreadformat to SLINEAR. It also
will place a write translation path on both channels from SLINEAR to the
channel's rawwriteformat. Using this option allows one to predictably set up
translation paths on all channels. There are two problems with this, though.
First and foremost, the transcode_via_sln option did not appear to be working
properly when I was placing a SIP call between two endpoints which did not
share any common formats. Second, even if this option were to work, for PLC
to be applied, there had to be a write translation path that would go from
some format to SLINEAR. It would not work properly if the starting format
of translation was SLINEAR.
The one-line change presented in this review request in chan_sip.c fixed the
first issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the format passed to
sip_request_call. This is nativeformats of the inbound channel. Because of this,
when ast_channel_make_compatible was called by app_dial, both channels already
had compatibly read and write formats. Thus, no translation path was set up at
the time. My change is to set the jointcapability of the sip_pvt created during
sip_request_call to the intersection of the inbound channel's nativeformats and
the configured peer capability that we determined during the earlier call to
create_addr. Doing this got the translation paths set up as expected when using
transcode_via_sln.
The changes presented in channel.c fixed the second issue for me. First and
foremost, when Asterisk is started, we'll read codecs.conf to see the value of
the genericplc option. If this option is set, and ast_write is called for a
frame with no data, then we will attempt to fill in the missing samples for
the frame. The implementation uses a channel datastore for maintaining the
PLC state and for creating a buffer to store PLC samples in. Even when we
receive a frame with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a basis for when
it comes time to actually do a PLC fill-in.
So, reviewers, now I ask for your help. First off, there's the one line change
in chan_sip that I have put in. Is it right? By my logic it seems correct, but
I'm sure someone can tell me why it is not going to work. This is probably the
change I'm least concerned about, though. What concerns me much more is the
set of changes in channel.c. First off, am I even doing it right? When I run
tests, I can clearly see that when PLC is activated, I see a significant increase
in RTP traffic where I would expect it to be. However, in my humble opinion, the
audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to
me than when no PLC is used at all. I need someone to review the logic I have used
to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic
is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm
sure someone can point out somewhere where I've done something incorrectly.
As I was writing this review request up, I decided to give the code a test run under
valgrind, and I find that for some reason, calls to plc_rx are causing some invalid
reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around
a bit to see why that is the case. If it's obvious to someone reviewing, speak up!
Finally, I have one other proposal that is not reflected in my code review. Since
without transcode_via_sln set, one cannot predict or control where a translation
path will be up, it seems to me that the current practice of using PLC only when
transcoding to SLINEAR is not useful. I recommend that once it has been determined
that the method used in this code review is correct and works as expected, then
the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 21:29:08 +00:00
if ( ast_opt_generic_plc & & fr - > subclass . codec = = AST_FORMAT_SLINEAR ) {
apply_plc ( chan , 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 */
2011-09-22 06:18:33 +00:00
if ( fr - > subclass . codec = = chan - > rawwriteformat ) {
2007-08-08 19:30:52 +00:00
f = fr ;
2011-09-22 06:18:33 +00:00
} else {
/* XXX Something is not right we are not compatible with this frame bad things can happen
* problems range from no / one - way audio to unexplained line hangups as a last resort try adjust the format
* ideally we do not want to do this and this indicates a deeper problem for now we log these events to
* eliminate user impact and help identify the problem areas
* JIRA issues related to this : -
* ASTERISK - 14384 , ASTERISK - 17502 , ASTERISK - 17541 , ASTERISK - 18063 , ASTERISK - 18325 , ASTERISK - 18422 */
if ( ( ! ( fr - > subclass . codec & chan - > nativeformats ) ) & & ( chan - > writeformat ! = fr - > subclass . codec ) ) {
char nf [ 512 ] ;
ast_log ( LOG_WARNING , " Codec mismatch on channel %s setting write format to %s from %s native formats %s \n " ,
chan - > name , ast_getformatname ( fr - > subclass . codec ) , ast_getformatname ( chan - > writeformat ) ,
ast_getformatname_multiple ( nf , sizeof ( nf ) , chan - > nativeformats & AST_FORMAT_AUDIO_MASK ) ) ;
ast_set_write_format ( chan , fr - > subclass . codec ) ;
}
2007-08-08 19:30:52 +00:00
f = ( chan - > writetrans ) ? ast_translate ( chan - > writetrans , fr , 0 ) : fr ;
2011-09-22 06:18:33 +00:00
}
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
2009-06-16 18:54:30 +00:00
if ( chan - > audiohooks ) {
2009-08-26 16:38:53 +00:00
struct ast_frame * prev = NULL , * new_frame , * cur , * dup ;
2009-11-06 22:35:44 +00:00
int freeoldlist = 0 ;
if ( f ! = fr ) {
freeoldlist = 1 ;
}
2009-06-16 18:54:30 +00:00
2009-08-26 16:38:53 +00:00
/* Since ast_audiohook_write may return a new frame, and the cur frame is
* an item in a list of frames , create a new list adding each cur frame back to it
* regardless if the cur frame changes or not . */
2009-06-16 18:54:30 +00:00
for ( cur = f ; cur ; cur = AST_LIST_NEXT ( cur , frame_list ) ) {
new_frame = ast_audiohook_write_list ( chan , chan - > audiohooks , AST_AUDIOHOOK_DIRECTION_WRITE , cur ) ;
2009-08-26 16:38:53 +00:00
/* if this frame is different than cur, preserve the end of the list,
* free the old frames , and set cur to be the new frame */
2009-06-16 18:54:30 +00:00
if ( new_frame ! = cur ) {
2009-11-06 22:35:44 +00:00
2009-08-26 16:38:53 +00:00
/* doing an ast_frisolate here seems silly, but we are not guaranteed the new_frame
* isn ' t part of local storage , meaning if ast_audiohook_write is called multiple
* times it may override the previous frame we got from it unless we dup it */
if ( ( dup = ast_frisolate ( new_frame ) ) ) {
AST_LIST_NEXT ( dup , frame_list ) = AST_LIST_NEXT ( cur , frame_list ) ;
2009-11-06 22:35:44 +00:00
if ( freeoldlist ) {
AST_LIST_NEXT ( cur , frame_list ) = NULL ;
ast_frfree ( cur ) ;
}
2010-07-29 19:13:27 +00:00
if ( new_frame ! = dup ) {
ast_frfree ( new_frame ) ;
}
2009-08-26 16:38:53 +00:00
cur = dup ;
}
}
/* now, regardless if cur is new or not, add it to the new list,
* if the new list has not started , cur will become the first item . */
if ( prev ) {
AST_LIST_NEXT ( prev , frame_list ) = cur ;
} else {
f = cur ; /* set f to be the beginning of our new list */
2009-06-16 18:54:30 +00:00
}
2009-08-26 16:38:53 +00:00
prev = cur ;
2009-06-16 18:54:30 +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 */
2009-06-16 18:54:30 +00:00
/* the translator on chan->writetrans may have returned multiple frames
from the single frame we passed in ; if so , feed each one of them to the
monitor */
2006-09-26 20:11:44 +00:00
if ( chan - > monitor & & chan - > monitor - > write_stream ) {
2009-06-16 18:54:30 +00:00
struct ast_frame * cur ;
for ( cur = f ; cur ; cur = AST_LIST_NEXT ( cur , frame_list ) ) {
2006-09-26 20:11:44 +00:00
/* XXX must explain this code */
2003-08-28 20:02:10 +00:00
# ifndef MONITOR_CONSTANT_DELAY
2009-06-16 18:54:30 +00:00
int jump = chan - > insmpl - chan - > outsmpl - 4 * cur - > samples ;
if ( jump > = 0 ) {
2010-02-16 17:07:41 +00:00
jump = calc_monitor_jump ( ( chan - > insmpl - chan - > outsmpl ) , ast_format_rate ( f - > subclass . codec ) , ast_format_rate ( chan - > monitor - > read_stream - > fmt - > format ) ) ;
2009-06-16 18:54:30 +00:00
if ( ast_seekstream ( chan - > monitor - > write_stream , jump , SEEK_FORCECUR ) = = - 1 )
ast_log ( LOG_WARNING , " Failed to perform seek in monitoring write stream, synchronization between the files may be broken \n " ) ;
2010-02-16 17:07:41 +00:00
chan - > outsmpl + = ( chan - > insmpl - chan - > outsmpl ) + cur - > samples ;
2009-06-16 18:54:30 +00:00
} else {
chan - > outsmpl + = cur - > samples ;
}
2003-08-28 20:02:10 +00:00
# else
2010-02-16 17:07:41 +00:00
int jump = calc_monitor_jump ( ( chan - > insmpl - chan - > outsmpl ) , ast_format_rate ( f - > subclass . codec ) , ast_format_rate ( chan - > monitor - > read_stream - > fmt - > format ) ) ;
2009-06-16 18:54:30 +00:00
if ( jump - MONITOR_DELAY > = 0 ) {
if ( ast_seekstream ( chan - > monitor - > write_stream , jump - cur - > samples , SEEK_FORCECUR ) = = - 1 )
ast_log ( LOG_WARNING , " Failed to perform seek in monitoring write stream, synchronization between the files may be broken \n " ) ;
2010-02-16 17:07:41 +00:00
chan - > outsmpl + = chan - > insmpl - chan - > outsmpl ;
2009-06-16 18:54:30 +00:00
} else {
chan - > outsmpl + = cur - > samples ;
}
2003-08-28 20:02:10 +00:00
# endif
2009-06-16 18:54:30 +00:00
if ( chan - > monitor - > state = = AST_MONITOR_RUNNING ) {
if ( ast_writestream ( chan - > monitor - > write_stream , cur ) < 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
2009-06-16 18:54:30 +00:00
/* the translator on chan->writetrans may have returned multiple frames
from the single frame we passed in ; if so , feed each one of them to the
channel , freeing each one after it has been written */
if ( ( f ! = fr ) & & AST_LIST_NEXT ( f , frame_list ) ) {
struct ast_frame * cur , * next ;
unsigned int skip = 0 ;
for ( cur = f , next = AST_LIST_NEXT ( cur , frame_list ) ;
cur ;
cur = next , next = cur ? AST_LIST_NEXT ( cur , frame_list ) : NULL ) {
if ( ! skip ) {
if ( ( res = chan - > tech - > write ( chan , cur ) ) < 0 ) {
chan - > _softhangup | = AST_SOFTHANGUP_DEV ;
skip = 1 ;
} else if ( next ) {
/* don't do this for the last frame in the list,
as the code outside the loop will do it once
*/
chan - > fout = FRAMECOUNT_INC ( chan - > fout ) ;
}
}
ast_frfree ( cur ) ;
}
/* reset f so the code below doesn't attempt to free it */
f = NULL ;
} else {
res = chan - > tech - > write ( chan , f ) ;
}
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break ;
case AST_FRAME_NULL :
case AST_FRAME_IAX :
/* Ignore these */
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res = 0 ;
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break ;
default :
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/* 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 ) ;
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break ;
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}
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2006-04-16 13:58:43 +00:00
if ( f & & f ! = fr )
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ast_frfree ( f ) ;
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ast_clear_flag ( chan , AST_FLAG_BLOCKING ) ;
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/* Consider a write failure to force a soft hangup */
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if ( res < 0 ) {
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chan - > _softhangup | = AST_SOFTHANGUP_DEV ;
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} else {
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chan - > fout = FRAMECOUNT_INC ( chan - > fout ) ;
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}
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done :
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if ( chan - > audiohooks & & ast_audiohook_write_list_empty ( chan - > audiohooks ) ) {
/* The list gets recreated if audiohooks are added again later */
ast_audiohook_detach_list ( chan - > audiohooks ) ;
chan - > audiohooks = NULL ;
}
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ast_channel_unlock ( chan ) ;
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return res ;
}
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static int set_format ( struct ast_channel * chan , format_t fmt , format_t * rawformat , format_t * format ,
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struct ast_trans_pvt * * trans , const int direction )
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{
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format_t native , native_fmt = ast_best_codec ( fmt ) ;
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int res ;
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char from [ 200 ] , to [ 200 ] ;
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/* Make sure we only consider audio */
fmt & = AST_FORMAT_AUDIO_MASK ;
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native = chan - > nativeformats ;
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if ( ! fmt | | ! native ) /* No audio requested */
return 0 ; /* Let's try a call without any sounds (video, text) */
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/* See if the underlying channel driver is capable of performing transcoding for us */
if ( ! ast_channel_setoption ( chan , direction ? AST_OPTION_FORMAT_WRITE : AST_OPTION_FORMAT_READ , & native_fmt , sizeof ( int * ) , 0 ) ) {
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ast_debug ( 1 , " Channel driver natively set channel %s to %s format %s \n " , chan - > name ,
direction ? " write " : " read " , ast_getformatname ( native_fmt ) ) ;
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chan - > nativeformats = * rawformat = * format = native_fmt ;
if ( * trans ) {
ast_translator_free_path ( * trans ) ;
}
* trans = NULL ;
return 0 ;
}
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/* 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 ) ;
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if ( res < 0 ) {
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ast_log ( LOG_WARNING , " Unable to find a codec translation path from %s to %s \n " ,
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ast_getformatname_multiple ( from , sizeof ( from ) , native ) ,
ast_getformatname_multiple ( to , sizeof ( to ) , fmt ) ) ;
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return - 1 ;
}
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/* Now we have a good choice for both. */
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ast_channel_lock ( chan ) ;
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if ( ( * rawformat = = native ) & & ( * format = = fmt ) & & ( ( * rawformat = = * format ) | | ( * trans ) ) ) {
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/* the channel is already in these formats, so nothing to do */
ast_channel_unlock ( chan ) ;
return 0 ;
}
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* rawformat = native ;
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/* User perspective is fmt */
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* format = fmt ;
/* Free any read translation we have right now */
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if ( * trans ) {
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ast_translator_free_path ( * trans ) ;
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* trans = NULL ;
}
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/* Build a translation path from the raw format to the desired format */
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if ( * format = = * rawformat ) {
/*
* If we were able to swap the native format to the format that
* has been requested , then there is no need to try to build
* a translation path .
*/
res = 0 ;
} else {
if ( ! direction ) {
/* reading */
* trans = ast_translator_build_path ( * format , * rawformat ) ;
} else {
/* writing */
* trans = ast_translator_build_path ( * rawformat , * format ) ;
}
res = * trans ? 0 : - 1 ;
}
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ast_channel_unlock ( chan ) ;
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ast_debug ( 1 , " Set channel %s to %s format %s \n " , chan - > name ,
direction ? " write " : " read " , ast_getformatname ( fmt ) ) ;
Merged revisions 296001 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines
Merged revisions 296000 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines
Handle failures building translation paths more effectively.
The problem scenario occurred on a heavily loaded system that was using the
codec_dahdi module and exceeded the hardware transcoding capacity. The failure
mode at that point was not good. The report came in to us as an Asterisk
lock-up. The "core show locks" shows a ton of threads locked up (but no
obvious deadlock). Upon deeper investigation, when the system is in this
state, the CPU was maxed out. The CPU was being consumed by the Asterisk
logger spewing messages on every audio frame for calls set up after transcoder
capacity was reached.
The purpose of this patch is to make Asterisk handle failures to create a
translation path in a more graceful manner. If we can't translate, then the
call just needs to be dropped, as it's not going to work. These are the
changes:
1) In set_format() of channel.c (which is called by set_read_format() and
set_write_format()), it was ignoring if ast_translator_build_path() failed and
returned NULL. It now pays attention to that case and returns a result
reflecting failure. With this change in place, the bridging code will
immediately detect a failure and end the bridge instead of proceeding to try to
bridge frames that can't be translated and making channel drivers freak out by
sending them frames in a format they weren't expecting.
2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was
ignored. It is now reflected in the return value of the function. This didn't
turn out to have any affect on the bug, but seemed like a good change to leave
in.
3) In app_dial(), when only sending a call to a single endpoint, it will
attempt to do some bridging of its own of early audio. It uses
make_compatible() when it's going to do this. However, it ignored failure from
make compatible. So, even with the fix from #1, if there was early audio going
through app_dial, there would still be a period of invalid frames passing
through. After detecting failure here, Dial() exits.
ABE-2658
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296002 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24 17:13:08 +00:00
return res ;
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}
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int ast_set_read_format ( struct ast_channel * chan , format_t fmt )
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{
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return set_format ( chan , fmt , & chan - > rawreadformat , & chan - > readformat ,
& chan - > readtrans , 0 ) ;
}
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int ast_set_write_format ( struct ast_channel * chan , format_t fmt )
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{
return set_format ( chan , fmt , & chan - > rawwriteformat , & chan - > writeformat ,
& chan - > writetrans , 1 ) ;
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}
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const char * ast_channel_reason2str ( int reason )
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{
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!! " ;
}
}
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static void handle_cause ( int cause , int * outstate )
{
if ( outstate ) {
/* compute error and return */
if ( cause = = AST_CAUSE_BUSY )
* outstate = AST_CONTROL_BUSY ;
else if ( cause = = AST_CAUSE_CONGESTION )
* outstate = AST_CONTROL_CONGESTION ;
else
* outstate = 0 ;
}
}
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/*!
* \ internal
* \ brief Helper function to inherit info from parent channel .
*
* \ param new_chan Channel inheriting information .
* \ param parent Channel new_chan inherits information .
* \ param orig Channel being replaced by the call forward channel .
*
* \ return Nothing
*/
static void call_forward_inherit ( struct ast_channel * new_chan , struct ast_channel * parent , struct ast_channel * orig )
{
if ( ! ast_test_flag ( parent , AST_FLAG_ZOMBIE ) & & ! ast_check_hangup ( parent ) ) {
struct ast_party_redirecting redirecting ;
/*
* The parent is not a ZOMBIE or hungup so update it with the
* original channel ' s redirecting information .
*/
ast_party_redirecting_init ( & redirecting ) ;
ast_channel_lock ( orig ) ;
ast_party_redirecting_copy ( & redirecting , & orig - > redirecting ) ;
ast_channel_unlock ( orig ) ;
if ( ast_channel_redirecting_macro ( orig , parent , & redirecting , 1 , 0 ) ) {
ast_channel_update_redirecting ( parent , & redirecting , NULL ) ;
}
ast_party_redirecting_free ( & redirecting ) ;
}
/* Safely inherit variables and datastores from the parent channel. */
ast_channel_lock_both ( parent , new_chan ) ;
ast_channel_inherit_variables ( parent , new_chan ) ;
ast_channel_datastore_inherit ( parent , new_chan ) ;
ast_channel_unlock ( new_chan ) ;
ast_channel_unlock ( parent ) ;
}
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struct ast_channel * ast_call_forward ( struct ast_channel * caller , struct ast_channel * orig , int * timeout , format_t format , struct outgoing_helper * oh , int * outstate )
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{
char tmpchan [ 256 ] ;
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struct ast_channel * new_chan = NULL ;
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char * data , * type ;
int cause = 0 ;
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int res ;
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/* gather data and request the new forward channel */
ast_copy_string ( tmpchan , orig - > call_forward , sizeof ( tmpchan ) ) ;
if ( ( data = strchr ( tmpchan , ' / ' ) ) ) {
* data + + = ' \0 ' ;
type = tmpchan ;
} else {
const char * forward_context ;
ast_channel_lock ( orig ) ;
forward_context = pbx_builtin_getvar_helper ( orig , " FORWARD_CONTEXT " ) ;
snprintf ( tmpchan , sizeof ( tmpchan ) , " %s@%s " , orig - > call_forward , S_OR ( forward_context , orig - > context ) ) ;
ast_channel_unlock ( orig ) ;
data = tmpchan ;
type = " Local " ;
}
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if ( ! ( new_chan = ast_request ( type , format , orig , data , & cause ) ) ) {
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ast_log ( LOG_NOTICE , " Unable to create channel for call forward to '%s/%s' (cause = %d) \n " , type , data , cause ) ;
handle_cause ( cause , outstate ) ;
ast_hangup ( orig ) ;
return NULL ;
}
/* Copy/inherit important information into new channel */
if ( oh ) {
if ( oh - > vars ) {
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ast_set_variables ( new_chan , oh - > vars ) ;
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}
if ( oh - > parent_channel ) {
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call_forward_inherit ( new_chan , oh - > parent_channel , orig ) ;
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}
if ( oh - > account ) {
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ast_channel_lock ( new_chan ) ;
ast_cdr_setaccount ( new_chan , oh - > account ) ;
ast_channel_unlock ( new_chan ) ;
2009-06-02 21:17:49 +00:00
}
} else if ( caller ) { /* no outgoing helper so use caller if avaliable */
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call_forward_inherit ( new_chan , caller , orig ) ;
2009-06-02 21:17:49 +00:00
}
2011-12-16 23:51:13 +00:00
ast_channel_lock_both ( orig , new_chan ) ;
ast_copy_flags ( new_chan - > cdr , orig - > cdr , AST_CDR_FLAG_ORIGINATED ) ;
ast_string_field_set ( new_chan , accountcode , orig - > accountcode ) ;
ast_party_connected_line_copy ( & new_chan - > connected , & orig - > connected ) ;
ast_party_redirecting_copy ( & new_chan - > redirecting , & orig - > redirecting ) ;
ast_channel_unlock ( new_chan ) ;
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ast_channel_unlock ( orig ) ;
/* call new channel */
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res = ast_call ( new_chan , data , 0 ) ;
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if ( timeout ) {
* timeout = res ;
}
if ( res ) {
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ast_log ( LOG_NOTICE , " Unable to call forward to channel %s/%s \n " , type , ( char * ) data ) ;
ast_hangup ( orig ) ;
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ast_hangup ( new_chan ) ;
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return NULL ;
}
ast_hangup ( orig ) ;
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return new_chan ;
2009-06-02 21:17:49 +00:00
}
2009-11-04 14:05:12 +00:00
struct ast_channel * __ast_request_and_dial ( const char * type , format_t format , const struct ast_channel * requestor , void * data , int timeout , int * outstate , const char * cid_num , const char * cid_name , struct outgoing_helper * oh )
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{
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int dummy_outstate ;
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int cause = 0 ;
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struct ast_channel * chan ;
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int res = 0 ;
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int last_subclass = 0 ;
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struct ast_party_connected_line connected ;
2005-07-18 05:00:23 +00:00
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if ( outstate )
* outstate = 0 ;
else
outstate = & dummy_outstate ; /* make outstate always a valid pointer */
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chan = ast_request ( type , format , requestor , data , & cause ) ;
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if ( ! chan ) {
ast_log ( LOG_NOTICE , " Unable to request channel %s/%s \n " , type , ( char * ) data ) ;
2009-06-02 21:17:49 +00:00
handle_cause ( cause , outstate ) ;
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return NULL ;
}
if ( oh ) {
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if ( oh - > vars ) {
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ast_set_variables ( chan , oh - > vars ) ;
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}
if ( ! ast_strlen_zero ( oh - > cid_num ) & & ! ast_strlen_zero ( oh - > cid_name ) ) {
/*
* Use the oh values instead of the function parameters for the
* outgoing CallerID .
*/
cid_num = oh - > cid_num ;
cid_name = oh - > cid_name ;
}
2007-03-03 14:40:18 +00:00
if ( oh - > parent_channel ) {
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/* Safely inherit variables and datastores from the parent channel. */
ast_channel_lock_both ( oh - > parent_channel , chan ) ;
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ast_channel_inherit_variables ( oh - > parent_channel , chan ) ;
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ast_channel_datastore_inherit ( oh - > parent_channel , chan ) ;
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ast_channel_unlock ( oh - > parent_channel ) ;
ast_channel_unlock ( chan ) ;
2007-03-03 14:40:18 +00:00
}
2011-12-16 20:55:17 +00:00
if ( oh - > account ) {
ast_channel_lock ( chan ) ;
ast_cdr_setaccount ( chan , oh - > account ) ;
ast_channel_unlock ( chan ) ;
}
2006-04-16 15:13:39 +00:00
}
2009-04-03 22:41:46 +00:00
2009-05-29 19:04:24 +00:00
ast_set_flag ( chan - > cdr , AST_CDR_FLAG_ORIGINATED ) ;
2009-04-03 22:41:46 +00:00
ast_party_connected_line_set_init ( & connected , & chan - > connected ) ;
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if ( cid_num ) {
connected . id . number . valid = 1 ;
connected . id . number . str = ( char * ) cid_num ;
connected . id . number . presentation = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED ;
}
if ( cid_name ) {
connected . id . name . valid = 1 ;
connected . id . name . str = ( char * ) cid_name ;
connected . id . name . presentation = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED ;
}
ast_channel_set_connected_line ( chan , & connected , NULL ) ;
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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 ) ;
2009-10-09 18:34:08 +00:00
if ( res = = 0 ) { /* timeout, treat it like ringing */
* outstate = AST_CONTROL_RINGING ;
break ;
}
if ( res < 0 ) /* error or done */
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break ;
if ( timeout > - 1 )
timeout = res ;
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if ( ! ast_strlen_zero ( chan - > call_forward ) ) {
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if ( ! ( chan = ast_call_forward ( NULL , chan , NULL , format , oh , outstate ) ) ) {
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return NULL ;
}
continue ;
}
2006-04-16 15:13:39 +00:00
f = ast_read ( chan ) ;
if ( ! f ) {
* outstate = AST_CONTROL_HANGUP ;
res = 0 ;
break ;
}
if ( f - > frametype = = AST_FRAME_CONTROL ) {
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switch ( f - > subclass . integer ) {
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case AST_CONTROL_RINGING : /* record but keep going */
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* outstate = f - > subclass . integer ;
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break ;
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case AST_CONTROL_BUSY :
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ast_cdr_busy ( chan - > cdr ) ;
* outstate = f - > subclass . integer ;
timeout = 0 ;
break ;
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case AST_CONTROL_INCOMPLETE :
ast_cdr_failed ( chan - > cdr ) ;
* outstate = AST_CONTROL_CONGESTION ;
timeout = 0 ;
break ;
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case AST_CONTROL_CONGESTION :
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ast_cdr_failed ( chan - > cdr ) ;
* outstate = f - > subclass . integer ;
timeout = 0 ;
break ;
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case AST_CONTROL_ANSWER :
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ast_cdr_answer ( chan - > cdr ) ;
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* outstate = f - > subclass . integer ;
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timeout = 0 ; /* trick to force exit from the while() */
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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 :
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case AST_CONTROL_SRCCHANGE :
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case AST_CONTROL_CONNECTED_LINE :
case AST_CONTROL_REDIRECTING :
Merge Call completion support into trunk.
From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:
1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
multiple calls to the same device. This proved to not be such a good idea
when implementing protocol-specific monitors, and so we ended up using one
monitor per-device per-call.
3. There are some configuration options which were conceived after the document
was written. These are documented in the ccss.conf.sample that is on this
review request.
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.
This implements CCBS and CCNR in several flavors.
First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.
Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:
* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
what is defined in the referenced draft.
* Implementation of the draft required support for SIP PUBLISH. I attempted to write
this in a generic-enough fashion such that if someone were to want to write PUBLISH
support for other event packages, such as dialog-state or presence, most of the effort
would be in writing callbacks specific to the event package.
* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
parser. The PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly. The rest of the
PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
code. In other words, while there is PIDF support here, it is not in any state
where it could easily be applied to other event packages as is.
Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.
Review: https://reviewboard.asterisk.org/r/523
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09 15:31:32 +00:00
case AST_CONTROL_CC :
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 :
2009-11-04 14:05:12 +00:00
ast_log ( LOG_NOTICE , " Don't know what to do with control frame %d \n " , f - > subclass . integer ) ;
2002-09-10 20:48:20 +00:00
}
2009-11-04 14:05:12 +00:00
last_subclass = f - > subclass . integer ;
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 ) {
2011-12-16 20:55:17 +00:00
ast_channel_lock ( chan ) ;
if ( AST_CONTROL_RINGING = = last_subclass ) {
2008-01-11 19:12:05 +00:00
chan - > hangupcause = AST_CAUSE_NO_ANSWER ;
2011-12-16 20:55:17 +00:00
}
if ( ! chan - > cdr & & ( chan - > cdr = ast_cdr_alloc ( ) ) ) {
2006-02-15 01:48:54 +00:00
ast_cdr_init ( chan - > cdr , chan ) ;
2011-12-16 20:55:17 +00:00
}
2003-09-12 16:51:35 +00:00
if ( chan - > cdr ) {
char tmp [ 256 ] ;
2011-12-16 20:55:17 +00:00
2006-04-16 15:13:39 +00:00
snprintf ( tmp , sizeof ( tmp ) , " %s/%s " , type , ( char * ) data ) ;
2011-12-16 20:55:17 +00:00
ast_cdr_setapp ( chan - > cdr , " Dial " , tmp ) ;
2003-09-12 16:51:35 +00:00
ast_cdr_update ( chan ) ;
ast_cdr_start ( chan - > cdr ) ;
ast_cdr_end ( chan - > cdr ) ;
/* If the cause wasn't handled properly */
2011-12-16 20:55:17 +00:00
if ( ast_cdr_disposition ( chan - > cdr , chan - > hangupcause ) ) {
2003-09-12 16:51:35 +00:00
ast_cdr_failed ( chan - > cdr ) ;
2011-12-16 20:55:17 +00:00
}
2006-02-15 01:48:54 +00:00
}
2011-12-16 20:55:17 +00:00
ast_channel_unlock ( chan ) ;
2003-09-12 16:51:35 +00:00
ast_hangup ( chan ) ;
chan = NULL ;
}
2002-09-10 20:48:20 +00:00
return chan ;
}
2009-11-04 14:05:12 +00:00
struct ast_channel * ast_request_and_dial ( const char * type , format_t format , const struct ast_channel * requestor , void * data , int timeout , int * outstate , const char * cidnum , const char * cidname )
2003-09-22 15:27:09 +00:00
{
2009-06-26 15:28:53 +00:00
return __ast_request_and_dial ( type , format , requestor , data , timeout , outstate , cidnum , cidname , NULL ) ;
2003-09-22 15:27:09 +00:00
}
2010-06-08 05:29:08 +00:00
static int set_security_requirements ( const struct ast_channel * requestor , struct ast_channel * out )
{
int ops [ 2 ] [ 2 ] = {
{ AST_OPTION_SECURE_SIGNALING , 0 } ,
{ AST_OPTION_SECURE_MEDIA , 0 } ,
} ;
int i ;
struct ast_channel * r = ( struct ast_channel * ) requestor ; /* UGLY */
struct ast_datastore * ds ;
if ( ! requestor | | ! out ) {
return 0 ;
}
ast_channel_lock ( r ) ;
if ( ( ds = ast_channel_datastore_find ( r , & secure_call_info , NULL ) ) ) {
struct ast_secure_call_store * encrypt = ds - > data ;
ops [ 0 ] [ 1 ] = encrypt - > signaling ;
ops [ 1 ] [ 1 ] = encrypt - > media ;
} else {
ast_channel_unlock ( r ) ;
return 0 ;
}
ast_channel_unlock ( r ) ;
for ( i = 0 ; i < 2 ; i + + ) {
if ( ops [ i ] [ 1 ] ) {
if ( ast_channel_setoption ( out , ops [ i ] [ 0 ] , & ops [ i ] [ 1 ] , sizeof ( ops [ i ] [ 1 ] ) , 0 ) ) {
/* We require a security feature, but the channel won't provide it */
return - 1 ;
}
} else {
/* We don't care if we can't clear the option on a channel that doesn't support it */
ast_channel_setoption ( out , ops [ i ] [ 0 ] , & ops [ i ] [ 1 ] , sizeof ( ops [ i ] [ 1 ] ) , 0 ) ;
}
}
return 0 ;
}
2009-11-04 14:05:12 +00:00
struct ast_channel * ast_request ( const char * type , format_t format , const struct ast_channel * requestor , 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 ;
2009-11-04 14:05:12 +00:00
format_t capabilities ;
format_t fmt ;
2001-03-12 03:04:51 +00:00
int res ;
2004-10-26 22:25:43 +00:00
int foo ;
2009-11-04 14:05:12 +00:00
format_t videoformat = format & AST_FORMAT_VIDEO_MASK ;
format_t 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
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
if ( AST_RWLIST_RDLOCK ( & backends ) ) {
ast_log ( LOG_WARNING , " Unable to lock technology backend list \n " ) ;
1999-12-04 21:35:07 +00:00
return NULL ;
}
2005-09-29 17:40:24 +00:00
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_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 ) {
2009-11-04 14:05:12 +00:00
char tmp1 [ 256 ] , tmp2 [ 256 ] ;
ast_log ( LOG_WARNING , " No translator path exists for channel type %s (native %s) to %s \n " , type ,
ast_getformatname_multiple ( tmp1 , sizeof ( tmp1 ) , chan - > tech - > capabilities ) ,
ast_getformatname_multiple ( tmp2 , sizeof ( tmp2 ) , format ) ) ;
2008-02-18 17:12:13 +00:00
* cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_UNLOCK ( & backends ) ;
2008-02-18 17:12:13 +00:00
return NULL ;
}
1999-12-04 21:35:07 +00:00
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_UNLOCK ( & backends ) ;
2005-09-29 17:40:24 +00:00
if ( ! chan - > tech - > requester )
return NULL ;
2009-06-26 15:28:53 +00:00
if ( ! ( c = chan - > tech - > requester ( type , capabilities | videoformat | textformat , requestor , data , cause ) ) )
2005-09-29 17:40:24 +00:00
return NULL ;
2009-06-26 15:28:53 +00:00
2010-06-08 05:29:08 +00:00
if ( set_security_requirements ( requestor , c ) ) {
ast_log ( LOG_WARNING , " Setting security requirements failed \n " ) ;
c = ast_channel_release ( c ) ;
* cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL ;
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 ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
AST_RWLIST_UNLOCK ( & backends ) ;
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 235635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) | 48 lines
Correct CDR dispositions for BUSY/FAILED
This patch is simple in that it reorders the disposition defines so that the fix
for issue 12946 works properly (the default CDR disposition was changed to
AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to
ensure all CDR records are written.
The side effects of CDR changes are scary, so I'm documenting the test cases
performed to attempt to catch any regressions. The following tests were all
performed using 1.4 rev 195881 vs head (235571) + patch:
A calls B
C calls B (busy)
Hangup C
Hangup A
(Both SIP and features)
A calls B
A blind transfers to C
Hangup C
(Both SIP and features)
A calls B
A attended transfers to C
Hangup C
A calls B
A attended transfers to C (SIP)
C blind transfers to A (features)
Hangup A
All of the test scenario CDRs matched.
The following tests were performed just with the patch to ensure proper operation
(with unanswered=yes):
exten =>s,1,Answer
exten =>s,n,ResetCDR(w)
exten =>s,n,ResetCDR(w)
exten =>s,1,ResetCDR(w)
exten =>s,n,ResetCDR(w)
(closes issue #16180)
Reported by: aatef
Patches:
bug16180.patch uploaded by jpeeler (license 325)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235660 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-18 22:51:37 +00:00
if ( chan - > cdr ) {
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: https://origsvn.digium.com/svn/asterisk/trunk@135821 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-05 23:45:32 +00:00
ast_set_flag ( chan - > cdr , AST_CDR_FLAG_DIALED ) ;
Merged revisions 235635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) | 48 lines
Correct CDR dispositions for BUSY/FAILED
This patch is simple in that it reorders the disposition defines so that the fix
for issue 12946 works properly (the default CDR disposition was changed to
AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to
ensure all CDR records are written.
The side effects of CDR changes are scary, so I'm documenting the test cases
performed to attempt to catch any regressions. The following tests were all
performed using 1.4 rev 195881 vs head (235571) + patch:
A calls B
C calls B (busy)
Hangup C
Hangup A
(Both SIP and features)
A calls B
A blind transfers to C
Hangup C
(Both SIP and features)
A calls B
A attended transfers to C
Hangup C
A calls B
A attended transfers to C (SIP)
C blind transfers to A (features)
Hangup A
All of the test scenario CDRs matched.
The following tests were performed just with the patch to ensure proper operation
(with unanswered=yes):
exten =>s,1,Answer
exten =>s,n,ResetCDR(w)
exten =>s,n,ResetCDR(w)
exten =>s,1,ResetCDR(w)
exten =>s,n,ResetCDR(w)
(closes issue #16180)
Reported by: aatef
Patches:
bug16180.patch uploaded by jpeeler (license 325)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235660 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-18 22:51:37 +00:00
}
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 ) ;
2009-04-03 16:47:27 +00:00
2010-01-07 20:42:27 +00:00
if ( res < = 0 ) {
2009-04-03 16:47:27 +00:00
return res ;
}
for ( ; ; ) {
struct ast_frame * fr ;
res = ast_waitfor ( chan , - 1 ) ;
if ( res < 0 | | ! ( fr = ast_read ( chan ) ) ) {
res = - 1 ;
break ;
}
2009-11-04 14:05:12 +00:00
if ( fr - > frametype = = AST_FRAME_CONTROL & & fr - > subclass . integer = = AST_CONTROL_TRANSFER ) {
2009-04-03 16:47:27 +00:00
enum ast_control_transfer * message = fr - > data . ptr ;
if ( * message = = AST_TRANSFER_SUCCESS ) {
res = 1 ;
} else {
res = - 1 ;
}
ast_frfree ( fr ) ;
break ;
}
ast_frfree ( fr ) ;
}
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
2010-09-09 20:11:06 +00:00
struct ast_silence_generator * silgen = NULL ;
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 ) ;
2010-09-09 20:11:06 +00:00
if ( ! silgen & & ast_opt_transmit_silence )
silgen = ast_channel_start_silence_generator ( c ) ;
2003-02-23 06:00:11 +00:00
usleep ( 1000 ) ;
if ( ! d )
d = ast_waitfordigit_full ( c , to , audiofd , ctrlfd ) ;
} else {
2010-09-09 20:11:06 +00:00
if ( ! silgen & & ast_opt_transmit_silence )
silgen = ast_channel_start_silence_generator ( c ) ;
2003-02-23 06:00:11 +00:00
d = ast_waitfordigit_full ( c , to , audiofd , ctrlfd ) ;
}
2010-09-09 20:11:06 +00:00
if ( d < 0 ) {
ast_channel_stop_silence_generator ( c , silgen ) ;
2009-03-03 23:21:18 +00:00
return AST_GETDATA_FAILED ;
2010-09-09 20:11:06 +00:00
}
2003-02-23 06:00:11 +00:00
if ( d = = 0 ) {
2009-03-03 23:21:18 +00:00
s [ pos ] = ' \0 ' ;
2010-09-09 20:11:06 +00:00
ast_channel_stop_silence_generator ( c , silgen ) ;
2009-03-03 23:21:18 +00:00
return AST_GETDATA_TIMEOUT ;
2003-02-23 06:00:11 +00:00
}
if ( d = = 1 ) {
2009-03-03 23:21:18 +00:00
s [ pos ] = ' \0 ' ;
2010-09-09 20:11:06 +00:00
ast_channel_stop_silence_generator ( c , silgen ) ;
2009-03-03 23:21:18 +00:00
return AST_GETDATA_INTERRUPTED ;
}
if ( strchr ( enders , d ) & & ( pos = = 0 ) ) {
s [ pos ] = ' \0 ' ;
2010-09-09 20:11:06 +00:00
ast_channel_stop_silence_generator ( c , silgen ) ;
2009-03-03 23:21:18 +00:00
return AST_GETDATA_EMPTY_END_TERMINATED ;
2003-02-23 06:00:11 +00:00
}
2009-03-03 23:21:18 +00:00
if ( ! strchr ( enders , d ) ) {
2003-02-23 06:00:11 +00:00
s [ pos + + ] = d ;
2009-03-03 23:21:18 +00:00
}
2003-02-23 06:00:11 +00:00
if ( strchr ( enders , d ) | | ( pos > = len ) ) {
2009-03-03 23:21:18 +00:00
s [ pos ] = ' \0 ' ;
2010-09-09 20:11:06 +00:00
ast_channel_stop_silence_generator ( c , silgen ) ;
2009-03-03 23:21:18 +00:00
return AST_GETDATA_COMPLETE ;
1999-12-04 21:35:07 +00:00
}
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
{
2009-11-04 14:05:12 +00:00
format_t src , dst ;
2010-07-22 14:58:01 +00:00
int use_slin ;
2005-04-04 03:28:38 +00:00
2009-04-09 16:19:35 +00:00
/* See if the channel driver can natively make these two channels compatible */
if ( from - > tech - > bridge & & from - > tech - > bridge = = to - > tech - > bridge & &
! ast_channel_setoption ( from , AST_OPTION_MAKE_COMPATIBLE , to , sizeof ( struct ast_channel * ) , 0 ) ) {
return 0 ;
}
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 ) {
2009-11-04 14:05:12 +00:00
ast_log ( LOG_WARNING , " No path to translate from %s to %s \n " , from - > name , to - > name ) ;
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
Fix transcode_via_sln option with SIP calls and improve PLC usage.
From reviewboard:
The problem here is a bit complex, so try to bear with me...
It was noticed by a Digium customer that generic PLC (as configured in
codecs.conf) did not appear to actually be having any sort of benefit when
packet loss was introduced on an RTP stream. I reproduced this issue myself
by streaming a file across an RTP stream and dropping approx. 5% of the
RTP packets. I saw no real difference between when PLC was enabled or disabled
when using wireshark to analyze the RTP streams.
After analyzing what was going on, it became clear that one of the problems
faced was that when running my tests, the translation paths were being set
up in such a way that PLC could not possibly work as expected. To illustrate,
if packets are lost on channel A's read stream, then we expect that PLC will
be applied to channel B's write stream. The problem is that generic PLC can
only be done when there is a translation path that moves from some codec to
SLINEAR. When I would run my tests, I found that every single time, read
and write translation paths would be set up on channel A instead of channel
B. There appeared to be no real way to predict which channel the translation
paths would be set up on.
This is where Kevin swooped in to let me know about the transcode_via_sln
option in asterisk.conf. It is supposed to work by placing a read translation
path on both channels from the channel's rawreadformat to SLINEAR. It also
will place a write translation path on both channels from SLINEAR to the
channel's rawwriteformat. Using this option allows one to predictably set up
translation paths on all channels. There are two problems with this, though.
First and foremost, the transcode_via_sln option did not appear to be working
properly when I was placing a SIP call between two endpoints which did not
share any common formats. Second, even if this option were to work, for PLC
to be applied, there had to be a write translation path that would go from
some format to SLINEAR. It would not work properly if the starting format
of translation was SLINEAR.
The one-line change presented in this review request in chan_sip.c fixed the
first issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the format passed to
sip_request_call. This is nativeformats of the inbound channel. Because of this,
when ast_channel_make_compatible was called by app_dial, both channels already
had compatibly read and write formats. Thus, no translation path was set up at
the time. My change is to set the jointcapability of the sip_pvt created during
sip_request_call to the intersection of the inbound channel's nativeformats and
the configured peer capability that we determined during the earlier call to
create_addr. Doing this got the translation paths set up as expected when using
transcode_via_sln.
The changes presented in channel.c fixed the second issue for me. First and
foremost, when Asterisk is started, we'll read codecs.conf to see the value of
the genericplc option. If this option is set, and ast_write is called for a
frame with no data, then we will attempt to fill in the missing samples for
the frame. The implementation uses a channel datastore for maintaining the
PLC state and for creating a buffer to store PLC samples in. Even when we
receive a frame with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a basis for when
it comes time to actually do a PLC fill-in.
So, reviewers, now I ask for your help. First off, there's the one line change
in chan_sip that I have put in. Is it right? By my logic it seems correct, but
I'm sure someone can tell me why it is not going to work. This is probably the
change I'm least concerned about, though. What concerns me much more is the
set of changes in channel.c. First off, am I even doing it right? When I run
tests, I can clearly see that when PLC is activated, I see a significant increase
in RTP traffic where I would expect it to be. However, in my humble opinion, the
audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to
me than when no PLC is used at all. I need someone to review the logic I have used
to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic
is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm
sure someone can point out somewhere where I've done something incorrectly.
As I was writing this review request up, I decided to give the code a test run under
valgrind, and I find that for some reason, calls to plc_rx are causing some invalid
reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around
a bit to see why that is the case. If it's obvious to someone reviewing, speak up!
Finally, I have one other proposal that is not reflected in my code review. Since
without transcode_via_sln set, one cannot predict or control where a translation
path will be up, it seems to me that the current practice of using PLC only when
transcoding to SLINEAR is not useful. I recommend that once it has been determined
that the method used in this code review is correct and works as expected, then
the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 21:29:08 +00:00
* transcoding is needed ; if desired , force transcode path
* to use SLINEAR between channels , but only if there is
* no direct conversion available . If generic PLC is
* desired , then transcoding via SLINEAR is a requirement
*/
2010-07-22 14:58:01 +00:00
use_slin = ( src = = AST_FORMAT_SLINEAR | | dst = = AST_FORMAT_SLINEAR ) ;
Fix transcode_via_sln option with SIP calls and improve PLC usage.
From reviewboard:
The problem here is a bit complex, so try to bear with me...
It was noticed by a Digium customer that generic PLC (as configured in
codecs.conf) did not appear to actually be having any sort of benefit when
packet loss was introduced on an RTP stream. I reproduced this issue myself
by streaming a file across an RTP stream and dropping approx. 5% of the
RTP packets. I saw no real difference between when PLC was enabled or disabled
when using wireshark to analyze the RTP streams.
After analyzing what was going on, it became clear that one of the problems
faced was that when running my tests, the translation paths were being set
up in such a way that PLC could not possibly work as expected. To illustrate,
if packets are lost on channel A's read stream, then we expect that PLC will
be applied to channel B's write stream. The problem is that generic PLC can
only be done when there is a translation path that moves from some codec to
SLINEAR. When I would run my tests, I found that every single time, read
and write translation paths would be set up on channel A instead of channel
B. There appeared to be no real way to predict which channel the translation
paths would be set up on.
This is where Kevin swooped in to let me know about the transcode_via_sln
option in asterisk.conf. It is supposed to work by placing a read translation
path on both channels from the channel's rawreadformat to SLINEAR. It also
will place a write translation path on both channels from SLINEAR to the
channel's rawwriteformat. Using this option allows one to predictably set up
translation paths on all channels. There are two problems with this, though.
First and foremost, the transcode_via_sln option did not appear to be working
properly when I was placing a SIP call between two endpoints which did not
share any common formats. Second, even if this option were to work, for PLC
to be applied, there had to be a write translation path that would go from
some format to SLINEAR. It would not work properly if the starting format
of translation was SLINEAR.
The one-line change presented in this review request in chan_sip.c fixed the
first issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the format passed to
sip_request_call. This is nativeformats of the inbound channel. Because of this,
when ast_channel_make_compatible was called by app_dial, both channels already
had compatibly read and write formats. Thus, no translation path was set up at
the time. My change is to set the jointcapability of the sip_pvt created during
sip_request_call to the intersection of the inbound channel's nativeformats and
the configured peer capability that we determined during the earlier call to
create_addr. Doing this got the translation paths set up as expected when using
transcode_via_sln.
The changes presented in channel.c fixed the second issue for me. First and
foremost, when Asterisk is started, we'll read codecs.conf to see the value of
the genericplc option. If this option is set, and ast_write is called for a
frame with no data, then we will attempt to fill in the missing samples for
the frame. The implementation uses a channel datastore for maintaining the
PLC state and for creating a buffer to store PLC samples in. Even when we
receive a frame with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a basis for when
it comes time to actually do a PLC fill-in.
So, reviewers, now I ask for your help. First off, there's the one line change
in chan_sip that I have put in. Is it right? By my logic it seems correct, but
I'm sure someone can tell me why it is not going to work. This is probably the
change I'm least concerned about, though. What concerns me much more is the
set of changes in channel.c. First off, am I even doing it right? When I run
tests, I can clearly see that when PLC is activated, I see a significant increase
in RTP traffic where I would expect it to be. However, in my humble opinion, the
audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to
me than when no PLC is used at all. I need someone to review the logic I have used
to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic
is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm
sure someone can point out somewhere where I've done something incorrectly.
As I was writing this review request up, I decided to give the code a test run under
valgrind, and I find that for some reason, calls to plc_rx are causing some invalid
reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around
a bit to see why that is the case. If it's obvious to someone reviewing, speak up!
Finally, I have one other proposal that is not reflected in my code review. Since
without transcode_via_sln set, one cannot predict or control where a translation
path will be up, it seems to me that the current practice of using PLC only when
transcoding to SLINEAR is not useful. I recommend that once it has been determined
that the method used in this code review is correct and works as expected, then
the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 21:29:08 +00:00
if ( ( src ! = dst ) & & ( ast_opt_generic_plc | | ast_opt_transcode_via_slin ) & &
2010-07-22 14:58:01 +00:00
( ast_translate_path_steps ( dst , src ) ! = 1 | | use_slin ) )
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 ) {
2009-11-04 14:05:12 +00:00
ast_log ( LOG_WARNING , " Unable to set read format on channel %s to %s \n " , from - > name , ast_getformatname ( 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 ) {
2009-11-04 14:05:12 +00:00
ast_log ( LOG_WARNING , " Unable to set write format on channel %s to %s \n " , to - > name , ast_getformatname ( 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
}
2010-11-09 16:55:32 +00:00
static int __ast_channel_masquerade ( struct ast_channel * original , struct ast_channel * clonechan , struct ast_datastore * xfer_ds )
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
2011-08-31 15:20:31 +00:00
for ( ; ; ) {
final_orig = original ;
final_clone = clonechan ;
2006-06-01 20:53:17 +00:00
2011-08-31 15:20:31 +00:00
ast_channel_lock_both ( original , clonechan ) ;
2006-07-10 21:07:48 +00:00
2011-08-31 15:20:31 +00:00
if ( ast_test_flag ( original , AST_FLAG_ZOMBIE )
| | ast_test_flag ( clonechan , AST_FLAG_ZOMBIE ) ) {
/* Zombies! Run! */
ast_log ( LOG_WARNING ,
" Can't setup masquerade. One or both channels is dead. (%s <-- %s) \n " ,
original - > name , clonechan - > name ) ;
2008-08-10 19:35:50 +00:00
ast_channel_unlock ( clonechan ) ;
2007-09-28 05:37:16 +00:00
ast_channel_unlock ( original ) ;
2011-08-31 15:20:31 +00:00
return - 1 ;
}
/*
* 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
*/
if ( original - > _bridge
& & ( original - > _bridge ! = ast_bridged_channel ( original ) )
& & ( original - > _bridge - > _bridge ! = original ) ) {
final_orig = original - > _bridge ;
}
if ( clonechan - > _bridge
& & ( clonechan - > _bridge ! = ast_bridged_channel ( clonechan ) )
& & ( clonechan - > _bridge - > _bridge ! = clonechan ) ) {
final_clone = clonechan - > _bridge ;
}
if ( final_clone - > tech - > get_base_channel
& & ( base = final_clone - > tech - > get_base_channel ( final_clone ) ) ) {
final_clone = base ;
2007-09-28 05:37:16 +00:00
}
2011-08-31 15:20:31 +00:00
if ( ( final_orig ! = original ) | | ( final_clone ! = clonechan ) ) {
/*
* 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 ) ) {
ast_channel_unlock ( clonechan ) ;
ast_channel_unlock ( original ) ;
/* Try again */
continue ;
}
if ( ast_channel_trylock ( final_clone ) ) {
ast_channel_unlock ( final_orig ) ;
ast_channel_unlock ( clonechan ) ;
ast_channel_unlock ( original ) ;
/* Try again */
continue ;
}
2008-08-10 19:35:50 +00:00
ast_channel_unlock ( clonechan ) ;
2007-09-28 05:37:16 +00:00
ast_channel_unlock ( original ) ;
2011-08-31 15:20:31 +00:00
original = final_orig ;
clonechan = final_clone ;
if ( ast_test_flag ( original , AST_FLAG_ZOMBIE )
| | ast_test_flag ( clonechan , AST_FLAG_ZOMBIE ) ) {
/* Zombies! Run! */
ast_log ( LOG_WARNING ,
" Can't setup masquerade. One or both channels is dead. (%s <-- %s) \n " ,
original - > name , clonechan - > name ) ;
ast_channel_unlock ( clonechan ) ;
ast_channel_unlock ( original ) ;
return - 1 ;
}
2006-07-10 21:07:48 +00:00
}
2011-08-31 15:20:31 +00:00
break ;
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 ) ;
2010-09-20 23:20:03 +00:00
if ( ! original - > masqr & & ! original - > masq & & ! clonechan - > masq & & ! clonechan - > masqr ) {
2008-08-10 19:35:50 +00:00
original - > masq = clonechan ;
clonechan - > masqr = original ;
2010-11-09 16:55:32 +00:00
if ( xfer_ds ) {
ast_channel_datastore_add ( original , xfer_ds ) ;
}
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 ;
2010-09-20 23:20:03 +00:00
} else if ( original - > masq ) {
ast_log ( LOG_WARNING , " %s is already going to masquerade as %s \n " ,
original - > masq - > name , original - > name ) ;
} else if ( original - > masqr ) {
/* not yet as a previously planned masq hasn't yet happened */
ast_log ( LOG_WARNING , " %s is already going to masquerade as %s \n " ,
original - > name , original - > masqr - > name ) ;
} else if ( clonechan - > masq ) {
ast_log ( LOG_WARNING , " %s is already going to masquerade as %s \n " ,
clonechan - > masq - > name , clonechan - > name ) ;
} else { /* (clonechan->masqr) */
ast_log ( LOG_WARNING , " %s is already going to masquerade as %s \n " ,
clonechan - > name , clonechan - > masqr - > name ) ;
2004-04-20 03:16:01 +00:00
}
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
}
2010-11-09 16:55:32 +00:00
int ast_channel_masquerade ( struct ast_channel * original , struct ast_channel * clone )
{
return __ast_channel_masquerade ( original , clone , NULL ) ;
}
/*!
* \ internal
* \ brief Copy the source connected line information to the destination for a transfer .
* \ since 1.8
*
* \ param dest Destination connected line
* \ param src Source connected line
*
* \ return Nothing
*/
static void party_connected_line_copy_transfer ( struct ast_party_connected_line * dest , const struct ast_party_connected_line * src )
{
struct ast_party_connected_line connected ;
connected = * ( ( struct ast_party_connected_line * ) src ) ;
connected . source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER ;
/* Make sure empty strings will be erased. */
if ( ! connected . id . name . str ) {
connected . id . name . str = " " ;
}
if ( ! connected . id . number . str ) {
connected . id . number . str = " " ;
}
if ( ! connected . id . subaddress . str ) {
connected . id . subaddress . str = " " ;
}
if ( ! connected . id . tag ) {
connected . id . tag = " " ;
}
ast_party_connected_line_copy ( dest , & connected ) ;
}
/*! Transfer masquerade connected line exchange data. */
struct xfer_masquerade_ds {
/*! New ID for the target of the transfer (Masquerade original channel) */
struct ast_party_connected_line target_id ;
/*! New ID for the transferee of the transfer (Masquerade clone channel) */
struct ast_party_connected_line transferee_id ;
/*! TRUE if the target call is held. (Masquerade original channel) */
int target_held ;
/*! TRUE if the transferee call is held. (Masquerade clone channel) */
int transferee_held ;
} ;
/*!
* \ internal
* \ brief Destroy the transfer connected line exchange datastore information .
* \ since 1.8
*
* \ param data The datastore payload to destroy .
*
* \ return Nothing
*/
static void xfer_ds_destroy ( void * data )
{
struct xfer_masquerade_ds * ds = data ;
ast_party_connected_line_free ( & ds - > target_id ) ;
ast_party_connected_line_free ( & ds - > transferee_id ) ;
ast_free ( ds ) ;
}
static const struct ast_datastore_info xfer_ds_info = {
. type = " xfer_colp " ,
. destroy = xfer_ds_destroy ,
} ;
int ast_channel_transfer_masquerade (
struct ast_channel * target_chan ,
const struct ast_party_connected_line * target_id ,
int target_held ,
struct ast_channel * transferee_chan ,
const struct ast_party_connected_line * transferee_id ,
int transferee_held )
{
struct ast_datastore * xfer_ds ;
struct xfer_masquerade_ds * xfer_colp ;
int res ;
xfer_ds = ast_datastore_alloc ( & xfer_ds_info , NULL ) ;
if ( ! xfer_ds ) {
return - 1 ;
}
xfer_colp = ast_calloc ( 1 , sizeof ( * xfer_colp ) ) ;
if ( ! xfer_colp ) {
ast_datastore_free ( xfer_ds ) ;
return - 1 ;
}
party_connected_line_copy_transfer ( & xfer_colp - > target_id , target_id ) ;
xfer_colp - > target_held = target_held ;
party_connected_line_copy_transfer ( & xfer_colp - > transferee_id , transferee_id ) ;
xfer_colp - > transferee_held = transferee_held ;
xfer_ds - > data = xfer_colp ;
res = __ast_channel_masquerade ( target_chan , transferee_chan , xfer_ds ) ;
if ( res ) {
ast_datastore_free ( xfer_ds ) ;
}
return res ;
}
2009-10-07 22:58:38 +00:00
/*! \brief this function simply changes the name of the channel and issues a manager_event
* with out unlinking and linking the channel from the ao2_container . This should
* only be used when the channel has already been unlinked from the ao2_container .
*/
static void __ast_change_name_nolink ( struct ast_channel * chan , const char * newname )
2003-02-05 21:18:09 +00:00
{
2009-11-13 20:42:03 +00:00
ast_manager_event ( chan , EVENT_FLAG_CALL , " Rename " , " Channel: %s \r \n Newname: %s \r \n Uniqueid: %s \r \n " , chan - > name , newname , chan - > uniqueid ) ;
2006-02-01 23:05:28 +00:00
ast_string_field_set ( chan , name , newname ) ;
2009-10-07 22:58:38 +00:00
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
2009-10-07 22:58:38 +00:00
void ast_change_name ( struct ast_channel * chan , const char * newname )
{
/* We must re-link, as the hash value will change here. */
ao2_unlink ( channels , chan ) ;
ast_channel_lock ( chan ) ;
__ast_change_name_nolink ( chan , newname ) ;
ast_channel_unlock ( chan ) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
ao2_link ( channels , chan ) ;
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
}
2009-06-26 15:28:53 +00:00
/* return the oldest of two linkedids. linkedid is derived from
uniqueid which is formed like this : [ systemname - ] ctime . seq
The systemname , and the dash are optional , followed by the epoch
time followed by an integer sequence . Note that this is not a
decimal number , since 1.2 is less than 1.11 in uniqueid land .
To compare two uniqueids , we parse out the integer values of the
time and the sequence numbers and compare them , with time trumping
sequence .
*/
static const char * oldest_linkedid ( const char * a , const char * b )
{
const char * satime , * saseq ;
const char * sbtime , * sbseq ;
const char * dash ;
unsigned int atime , aseq , btime , bseq ;
if ( ast_strlen_zero ( a ) )
return b ;
if ( ast_strlen_zero ( b ) )
return a ;
satime = a ;
sbtime = b ;
/* jump over the system name */
if ( ( dash = strrchr ( satime , ' - ' ) ) ) {
satime = dash + 1 ;
}
if ( ( dash = strrchr ( sbtime , ' - ' ) ) ) {
sbtime = dash + 1 ;
}
/* the sequence comes after the '.' */
saseq = strchr ( satime , ' . ' ) ;
sbseq = strchr ( sbtime , ' . ' ) ;
if ( ! saseq | | ! sbseq )
return NULL ;
saseq + + ;
sbseq + + ;
/* convert it all to integers */
atime = atoi ( satime ) ; /* note that atoi is ignoring the '.' after the time string */
btime = atoi ( sbtime ) ; /* note that atoi is ignoring the '.' after the time string */
aseq = atoi ( saseq ) ;
bseq = atoi ( sbseq ) ;
/* and finally compare */
if ( atime = = btime ) {
return ( aseq < bseq ) ? a : b ;
}
else {
return ( atime < btime ) ? a : b ;
}
}
/*! Set the channel's linkedid to the given string, and also check to
* see if the channel ' s old linkedid is now being retired */
static void ast_channel_change_linkedid ( struct ast_channel * chan , const char * linkedid )
{
/* if the linkedid for this channel is being changed from something, check... */
if ( ! ast_strlen_zero ( chan - > linkedid ) & & 0 ! = strcmp ( chan - > linkedid , linkedid ) ) {
ast_cel_check_retire_linkedid ( chan ) ;
}
ast_string_field_set ( chan , linkedid , linkedid ) ;
}
/*!
\ brief Propagate the oldest linkedid between associated channels
*/
void ast_channel_set_linkgroup ( struct ast_channel * chan , struct ast_channel * peer )
{
const char * linkedid = NULL ;
struct ast_channel * bridged ;
linkedid = oldest_linkedid ( chan - > linkedid , peer - > linkedid ) ;
linkedid = oldest_linkedid ( linkedid , chan - > uniqueid ) ;
linkedid = oldest_linkedid ( linkedid , peer - > uniqueid ) ;
if ( chan - > _bridge ) {
bridged = ast_bridged_channel ( chan ) ;
if ( bridged ! = peer ) {
linkedid = oldest_linkedid ( linkedid , bridged - > linkedid ) ;
linkedid = oldest_linkedid ( linkedid , bridged - > uniqueid ) ;
}
}
if ( peer - > _bridge ) {
bridged = ast_bridged_channel ( peer ) ;
if ( bridged ! = chan ) {
linkedid = oldest_linkedid ( linkedid , bridged - > linkedid ) ;
linkedid = oldest_linkedid ( linkedid , bridged - > uniqueid ) ;
}
}
/* just in case setting a stringfield to itself causes problems */
linkedid = ast_strdupa ( linkedid ) ;
ast_channel_change_linkedid ( chan , linkedid ) ;
ast_channel_change_linkedid ( peer , linkedid ) ;
if ( chan - > _bridge ) {
bridged = ast_bridged_channel ( chan ) ;
if ( bridged ! = peer ) {
ast_channel_change_linkedid ( bridged , linkedid ) ;
}
}
if ( peer - > _bridge ) {
bridged = ast_bridged_channel ( peer ) ;
if ( bridged ! = chan ) {
ast_channel_change_linkedid ( bridged , linkedid ) ;
}
}
}
/* copy accountcode and peeraccount across during a link */
static void ast_set_owners_and_peers ( struct ast_channel * chan1 ,
struct ast_channel * chan2 )
{
if ( ! ast_strlen_zero ( chan1 - > accountcode ) & & ast_strlen_zero ( chan2 - > peeraccount ) ) {
ast_log ( LOG_DEBUG , " setting peeraccount to %s for %s from data on channel %s \n " ,
chan1 - > accountcode , chan2 - > name , chan1 - > name ) ;
ast_string_field_set ( chan2 , peeraccount , chan1 - > accountcode ) ;
}
if ( ! ast_strlen_zero ( chan2 - > accountcode ) & & ast_strlen_zero ( chan1 - > peeraccount ) ) {
ast_log ( LOG_DEBUG , " setting peeraccount to %s for %s from data on channel %s \n " ,
chan2 - > accountcode , chan1 - > name , chan2 - > name ) ;
ast_string_field_set ( chan1 , peeraccount , chan2 - > accountcode ) ;
}
if ( ! ast_strlen_zero ( chan1 - > peeraccount ) & & ast_strlen_zero ( chan2 - > accountcode ) ) {
ast_log ( LOG_DEBUG , " setting accountcode to %s for %s from data on channel %s \n " ,
chan1 - > peeraccount , chan2 - > name , chan1 - > name ) ;
ast_string_field_set ( chan2 , accountcode , chan1 - > peeraccount ) ;
}
if ( ! ast_strlen_zero ( chan2 - > peeraccount ) & & ast_strlen_zero ( chan1 - > accountcode ) ) {
ast_log ( LOG_DEBUG , " setting accountcode to %s for %s from data on channel %s \n " ,
chan2 - > peeraccount , chan1 - > name , chan2 - > name ) ;
ast_string_field_set ( chan1 , accountcode , chan2 - > peeraccount ) ;
}
if ( 0 ! = strcmp ( chan1 - > accountcode , chan2 - > peeraccount ) ) {
ast_log ( LOG_DEBUG , " changing peeraccount from %s to %s on %s to match channel %s \n " ,
chan2 - > peeraccount , chan1 - > peeraccount , chan2 - > name , chan1 - > name ) ;
ast_string_field_set ( chan2 , peeraccount , chan1 - > accountcode ) ;
}
if ( 0 ! = strcmp ( chan2 - > accountcode , chan1 - > peeraccount ) ) {
ast_log ( LOG_DEBUG , " changing peeraccount from %s to %s on %s to match channel %s \n " ,
chan1 - > peeraccount , chan2 - > peeraccount , chan1 - > name , chan2 - > name ) ;
ast_string_field_set ( chan1 , peeraccount , chan2 - > accountcode ) ;
}
}
2009-06-22 20:11:04 +00:00
/*!
* \ pre chan is locked
*/
2009-11-13 20:42:03 +00:00
static void report_new_callerid ( struct ast_channel * chan )
2009-06-22 20:11:04 +00:00
{
2010-07-14 15:48:36 +00:00
int pres ;
pres = ast_party_id_presentation ( & chan - > caller . id ) ;
2009-11-13 20:42:03 +00:00
ast_manager_event ( chan , EVENT_FLAG_CALL , " NewCallerid " ,
2010-07-14 15:48:36 +00:00
" Channel: %s \r \n "
" CallerIDNum: %s \r \n "
" CallerIDName: %s \r \n "
" Uniqueid: %s \r \n "
" CID-CallingPres: %d (%s) \r \n " ,
chan - > name ,
S_COR ( chan - > caller . id . number . valid , chan - > caller . id . number . str , " " ) ,
S_COR ( chan - > caller . id . name . valid , chan - > caller . id . name . str , " " ) ,
chan - > uniqueid ,
pres ,
ast_describe_caller_presentation ( pres )
) ;
2009-06-22 20:11:04 +00:00
}
2005-12-31 16:35:48 +00:00
/*!
2010-11-09 16:55:32 +00:00
* \ internal
* \ brief Transfer COLP between target and transferee channels .
* \ since 1.8
*
* \ param transferee Transferee channel to exchange connected line information .
* \ param colp Connected line information to exchange .
*
* \ return Nothing
*/
static void masquerade_colp_transfer ( struct ast_channel * transferee , struct xfer_masquerade_ds * colp )
{
struct ast_control_read_action_payload * frame_payload ;
int payload_size ;
int frame_size ;
unsigned char connected_line_data [ 1024 ] ;
2005-12-31 16:35:48 +00:00
2010-11-09 16:55:32 +00:00
/* Release any hold on the target. */
if ( colp - > target_held ) {
ast_queue_control ( transferee , AST_CONTROL_UNHOLD ) ;
}
/*
* Since transferee may not actually be bridged to another channel ,
* there is no way for us to queue a frame so that its connected
* line status will be updated . Instead , we use the somewhat
* hackish approach of using a special control frame type that
* instructs ast_read ( ) to perform a specific action . In this
* case , the frame we queue tells ast_read ( ) to call the
* connected line interception macro configured for transferee .
*/
payload_size = ast_connected_line_build_data ( connected_line_data ,
sizeof ( connected_line_data ) , & colp - > target_id , NULL ) ;
if ( payload_size ! = - 1 ) {
frame_size = payload_size + sizeof ( * frame_payload ) ;
frame_payload = alloca ( frame_size ) ;
frame_payload - > action = AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO ;
frame_payload - > payload_size = payload_size ;
memcpy ( frame_payload - > payload , connected_line_data , payload_size ) ;
ast_queue_control_data ( transferee , AST_CONTROL_READ_ACTION , frame_payload ,
frame_size ) ;
}
/*
* In addition to queueing the read action frame so that the
* connected line info on transferee will be updated , we also are
* going to queue a plain old connected line update on transferee to
* update the target .
*/
ast_channel_queue_connected_line_update ( transferee , & colp - > transferee_id , NULL ) ;
}
/*!
* \ brief Masquerade a channel
*
* \ note Assumes _NO_ channels and _NO_ channel pvt ' s are locked . If a channel is locked while calling
* this function , it invalidates our channel container locking order . All channels
* must be unlocked before it is permissible to lock the channels ' ao2 container .
*/
2004-04-06 22:17:32 +00:00
int ast_do_masquerade ( struct ast_channel * original )
2001-05-07 00:43:32 +00:00
{
2009-11-04 14:05:12 +00:00
format_t x ;
int i ;
2001-05-07 00:43:32 +00:00
int res = 0 ;
2004-07-09 10:57:43 +00:00
int origstate ;
2010-10-22 15:47:08 +00:00
int visible_indication ;
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 ;
2009-04-03 22:41:46 +00:00
union {
2010-07-14 15:48:36 +00:00
struct ast_party_dialed dialed ;
struct ast_party_caller caller ;
2009-04-03 22:41:46 +00:00
struct ast_party_connected_line connected ;
struct ast_party_redirecting redirecting ;
} exchange ;
2009-11-13 20:42:03 +00:00
struct ast_channel * clonechan , * chans [ 2 ] ;
2010-08-16 17:53:44 +00:00
struct ast_channel * bridged ;
2007-03-30 14:37:21 +00:00
struct ast_cdr * cdr ;
2010-11-09 16:55:32 +00:00
struct ast_datastore * xfer_ds ;
struct xfer_masquerade_ds * xfer_colp ;
2009-11-04 14:05:12 +00:00
format_t rformat = original - > readformat ;
format_t 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
2009-08-28 20:14:39 +00:00
/* XXX This operation is a bit odd. We're essentially putting the guts of
* the clone channel into the original channel . Start by killing off the
* original channel ' s backend . While the features are nice , which is the
* reason we ' re keeping it , it ' s still awesomely weird . XXX */
2001-05-07 00:43:32 +00:00
2009-10-07 22:58:38 +00:00
/* The reasoning for the channels ao2_container lock here is complex.
*
* In order to check for a race condition , the original channel must
* be locked . If it is determined that the masquerade should proceed
* the original channel can absolutely not be unlocked until the end
* of the function . Since after determining the masquerade should
* continue requires the channels to be unlinked from the ao2_container ,
* the container lock must be held first to achieve proper locking order .
*/
ao2_lock ( channels ) ;
2011-07-30 23:56:29 +00:00
/* lock the original channel to determine if the masquerade is required or not */
2009-10-07 22:58:38 +00:00
ast_channel_lock ( original ) ;
2011-07-30 23:56:29 +00:00
/*
* This checks to see if the masquerade has already happened or
* not . There is a race condition that exists for this
* function . Since all pvt and channel locks must be let go
* before calling do_masquerade , it is possible that it could be
* called multiple times for the same channel . This check
* verifies whether or not the masquerade has already been
* completed by another thread .
*/
2011-07-30 15:25:16 +00:00
while ( ( clonechan = original - > masq ) & & ast_channel_trylock ( clonechan ) ) {
2011-07-30 23:56:29 +00:00
/*
* A masq is needed but we could not get the clonechan lock
* immediately . Since this function already holds the global
* container lock , unlocking original for deadlock avoidance
* will not result in any sort of masquerade race condition . If
* masq is called by a different thread while this happens , it
* will be stuck waiting until we unlock the container .
*/
2009-10-07 22:58:38 +00:00
CHANNEL_DEADLOCK_AVOIDANCE ( original ) ;
}
2011-07-30 23:56:29 +00:00
/*
* A final masq check must be done after deadlock avoidance for
* clonechan above or we could get a double masq . This is
* posible with ast_hangup at least .
*/
if ( ! clonechan ) {
/* masq already completed by another thread, or never needed to be done to begin with */
2011-07-30 15:25:16 +00:00
ast_channel_unlock ( original ) ;
ao2_unlock ( channels ) ;
2011-07-30 23:56:29 +00:00
return 0 ;
2011-07-30 15:25:16 +00:00
}
2010-11-09 16:55:32 +00:00
/* Get any transfer masquerade connected line exchange data. */
xfer_ds = ast_channel_datastore_find ( original , & xfer_ds_info , NULL ) ;
if ( xfer_ds ) {
ast_channel_datastore_remove ( original , xfer_ds ) ;
xfer_colp = xfer_ds - > data ;
} else {
xfer_colp = NULL ;
}
/*
* Release any hold on the transferee channel before proceeding
* with the masquerade .
*/
if ( xfer_colp & & xfer_colp - > transferee_held ) {
ast_indicate ( clonechan , AST_CONTROL_UNHOLD ) ;
}
2009-10-07 22:58:38 +00:00
/* clear the masquerade channels */
original - > masq = NULL ;
clonechan - > masqr = NULL ;
/* unlink from channels container as name (which is the hash value) will change */
ao2_unlink ( channels , original ) ;
ao2_unlink ( channels , clonechan ) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
ast_debug ( 4 , " Actually Masquerading %s(%d) into the structure of %s(%d) \n " ,
clonechan - > name , clonechan - > _state , original - > name , original - > _state ) ;
2010-10-22 15:47:08 +00:00
/*
* Stop any visible indiction on the original channel so we can
* transfer it to the clonechan taking the original ' s place .
*/
visible_indication = original - > visible_indication ;
ast_indicate ( original , - 1 ) ;
2009-11-13 20:42:03 +00:00
chans [ 0 ] = clonechan ;
chans [ 1 ] = original ;
2010-10-22 15:47:08 +00:00
ast_manager_event_multichan ( EVENT_FLAG_CALL , " Masquerade " , 2 , chans ,
" Clone: %s \r \n "
" CloneState: %s \r \n "
" Original: %s \r \n "
" OriginalState: %s \r \n " ,
clonechan - > name , ast_state2str ( clonechan - > _state ) , original - > name , ast_state2str ( original - > _state ) ) ;
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 ) ;
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 ) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
2001-05-07 00:43:32 +00:00
/* Mangle the name of the clone channel */
2009-10-07 22:58:38 +00:00
__ast_change_name_nolink ( clonechan , masqn ) ;
2001-05-07 00:43:32 +00:00
2009-08-03 14:29:17 +00:00
/* Copy the name from the clone channel */
2009-10-07 22:58:38 +00:00
__ast_change_name_nolink ( original , newn ) ;
2009-08-03 14:29:17 +00:00
2009-06-26 15:28:53 +00:00
/* share linked id's */
ast_channel_set_linkgroup ( original , clonechan ) ;
/* 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
2009-10-07 22:58:38 +00:00
if ( clonechan - > tech - > fixup & & clonechan - > tech - > fixup ( original , clonechan ) ) {
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 */
2009-10-07 22:58:38 +00:00
if ( clonechan - > tech - > hangup & & clonechan - > tech - > hangup ( clonechan ) ) {
2001-05-07 00:43:32 +00:00
ast_log ( LOG_WARNING , " Hangup failed! Strange things may happen! \n " ) ;
2009-10-07 22:58:38 +00:00
res = - 1 ;
goto done ;
2001-05-07 00:43:32 +00:00
}
2008-01-03 21:27:57 +00:00
2011-05-25 16:23:11 +00:00
/*
* We just hung up the physical side of the channel . Set the
* new zombie to use the kill channel driver for safety .
*/
clonechan - > tech = & ast_kill_tech ;
2001-05-07 00:43:32 +00:00
/* Mangle the name of the clone channel */
2009-10-07 22:58:38 +00:00
snprintf ( zombn , sizeof ( zombn ) , " %s<ZOMBIE> " , orig ) ; /* quick, hide the brains! */
__ast_change_name_nolink ( clonechan , zombn ) ;
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 ;
2009-06-17 20:04:12 +00:00
/* We use a safe traversal here because some fixup routines actually
* remove the datastore from the list and free them .
*/
AST_LIST_TRAVERSE_SAFE_BEGIN ( & clonechan - > datastores , ds , entry ) {
2008-03-13 21:40:43 +00:00
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
}
2009-06-17 20:04:12 +00:00
AST_LIST_TRAVERSE_SAFE_END ;
2008-12-19 22:26:16 +00:00
AST_LIST_APPEND_LIST ( & original - > datastores , & clonechan - > datastores , entry ) ;
2008-03-13 21:40:43 +00:00
}
2006-04-10 23:29:50 +00:00
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
ast_autochan_new_channel ( clonechan , original ) ;
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 */
2010-01-14 18:03:31 +00:00
ast_set_flag ( original , ast_test_flag ( clonechan , AST_FLAG_EXCEPTION | AST_FLAG_OUTGOING ) ) ;
2008-08-10 19:35:50 +00:00
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
2009-04-03 22:41:46 +00:00
/*
* Just swap the whole structures , nevermind the allocations ,
* they ' ll work themselves out .
*/
2010-07-14 15:48:36 +00:00
exchange . dialed = original - > dialed ;
original - > dialed = clonechan - > dialed ;
clonechan - > dialed = exchange . dialed ;
exchange . caller = original - > caller ;
original - > caller = clonechan - > caller ;
clonechan - > caller = exchange . caller ;
2009-06-22 20:11:04 +00:00
2009-04-03 22:41:46 +00:00
exchange . connected = original - > connected ;
original - > connected = clonechan - > connected ;
clonechan - > connected = exchange . connected ;
2010-07-14 15:48:36 +00:00
2009-04-03 22:41:46 +00:00
exchange . redirecting = original - > redirecting ;
original - > redirecting = clonechan - > redirecting ;
clonechan - > redirecting = exchange . redirecting ;
2008-01-03 21:27:57 +00:00
2010-07-14 15:48:36 +00:00
report_new_callerid ( original ) ;
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
2009-06-26 15:28:53 +00:00
/* copy over accuntcode and set peeraccount across the bridge */
ast_string_field_set ( original , accountcode , S_OR ( clonechan - > accountcode , " " ) ) ;
if ( original - > _bridge ) {
/* XXX - should we try to lock original->_bridge here? */
ast_string_field_set ( original - > _bridge , peeraccount , S_OR ( clonechan - > accountcode , " " ) ) ;
ast_cel_report_event ( original , AST_CEL_BRIDGE_UPDATE , NULL , NULL , NULL ) ;
}
2009-11-04 14:05:12 +00:00
ast_debug ( 1 , " Putting channel %s in %s/%s formats \n " , original - > name ,
ast_getformatname ( wformat ) , ast_getformatname ( 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 ) {
2009-10-07 22:58:38 +00:00
if ( original - > tech - > fixup ( clonechan , original ) ) {
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 ) ;
2009-10-07 22:58:38 +00:00
res = - 1 ;
goto done ;
2001-05-07 00:43:32 +00:00
}
} 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
2008-12-15 14:40:24 +00:00
/*
* If an indication is currently playing , maintain it on the channel
* that is taking the place of original
*
* This is needed because the masquerade is swapping out in the internals
* of this channel , and the new channel private data needs to be made
* aware of the current visible indication ( RINGING , CONGESTION , etc . )
*/
2010-10-22 15:47:08 +00:00
if ( visible_indication ) {
ast_indicate ( original , visible_indication ) ;
2008-12-15 14:40:24 +00:00
}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +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 ) ;
2009-11-13 20:42:03 +00:00
ast_manager_event ( clonechan , 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
) ;
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
clonechan = ast_channel_release ( 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 ) ;
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 ) ;
2009-10-07 22:58:38 +00:00
2010-08-16 17:53:44 +00:00
if ( ( bridged = ast_bridged_channel ( original ) ) ) {
ast_channel_lock ( bridged ) ;
ast_indicate ( bridged , AST_CONTROL_SRCCHANGE ) ;
ast_channel_unlock ( bridged ) ;
}
ast_indicate ( original , AST_CONTROL_SRCCHANGE ) ;
2010-11-09 16:55:32 +00:00
if ( xfer_colp ) {
/*
* After the masquerade , the original channel pointer actually
* points to the new transferee channel and the bridged channel
* is still the intended transfer target party .
*/
masquerade_colp_transfer ( original , xfer_colp ) ;
}
2009-10-07 22:58:38 +00:00
done :
2010-11-09 16:55:32 +00:00
if ( xfer_ds ) {
ast_datastore_free ( xfer_ds ) ;
}
2009-10-07 22:58:38 +00:00
/* it is possible for the clone channel to disappear during this */
if ( clonechan ) {
ast_channel_unlock ( original ) ;
ast_channel_unlock ( clonechan ) ;
ao2_link ( channels , clonechan ) ;
ao2_link ( channels , original ) ;
} else {
ast_channel_unlock ( original ) ;
ao2_link ( channels , original ) ;
}
2010-08-16 17:53:44 +00:00
2010-09-20 22:21:50 +00:00
ao2_unlock ( channels ) ;
2010-11-09 22:46:45 +00:00
return res ;
2002-09-10 20:48:20 +00:00
}
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 ) {
2010-07-14 15:48:36 +00:00
chan - > caller . id . number . valid = 1 ;
ast_free ( chan - > caller . id . number . str ) ;
chan - > caller . id . number . str = ast_strdup ( cid_num ) ;
2004-10-02 00:58:31 +00:00
}
2007-11-25 19:33:33 +00:00
if ( cid_name ) {
2010-07-14 15:48:36 +00:00
chan - > caller . id . name . valid = 1 ;
ast_free ( chan - > caller . id . name . str ) ;
chan - > caller . id . name . str = ast_strdup ( cid_name ) ;
2004-10-02 00:58:31 +00:00
}
2007-11-25 19:33:33 +00:00
if ( cid_ani ) {
2010-07-14 16:58:03 +00:00
chan - > caller . ani . number . valid = 1 ;
ast_free ( chan - > caller . ani . number . str ) ;
chan - > caller . ani . number . str = ast_strdup ( cid_ani ) ;
2003-02-05 21:18:09 +00:00
}
2010-09-29 15:04:56 +00:00
if ( chan - > cdr ) {
ast_cdr_setcid ( chan - > cdr , chan ) ;
}
2009-06-22 20:11:04 +00:00
report_new_callerid ( chan ) ;
2007-11-29 00:28:10 +00:00
ast_channel_unlock ( chan ) ;
2002-09-10 20:48:20 +00:00
}
2010-07-14 15:48:36 +00:00
void ast_channel_set_caller ( struct ast_channel * chan , const struct ast_party_caller * caller , const struct ast_set_party_caller * update )
{
if ( & chan - > caller = = caller ) {
/* Don't set to self */
return ;
}
ast_channel_lock ( chan ) ;
ast_party_caller_set ( & chan - > caller , caller , update ) ;
ast_channel_unlock ( chan ) ;
}
void ast_channel_set_caller_event ( struct ast_channel * chan , const struct ast_party_caller * caller , const struct ast_set_party_caller * update )
{
2011-02-24 21:38:41 +00:00
const char * pre_set_number ;
const char * pre_set_name ;
2010-07-14 15:48:36 +00:00
if ( & chan - > caller = = caller ) {
/* Don't set to self */
return ;
}
ast_channel_lock ( chan ) ;
2011-02-24 21:38:41 +00:00
pre_set_number =
S_COR ( chan - > caller . id . number . valid , chan - > caller . id . number . str , NULL ) ;
pre_set_name = S_COR ( chan - > caller . id . name . valid , chan - > caller . id . name . str , NULL ) ;
2010-07-14 15:48:36 +00:00
ast_party_caller_set ( & chan - > caller , caller , update ) ;
2011-02-24 21:38:41 +00:00
if ( S_COR ( chan - > caller . id . number . valid , chan - > caller . id . number . str , NULL )
! = pre_set_number
| | S_COR ( chan - > caller . id . name . valid , chan - > caller . id . name . str , NULL )
! = pre_set_name ) {
2010-07-14 15:48:36 +00:00
/* The caller id name or number changed. */
report_new_callerid ( chan ) ;
}
2010-09-29 17:08:20 +00:00
if ( chan - > cdr ) {
ast_cdr_setcid ( chan - > cdr , chan ) ;
}
2010-07-14 15:48:36 +00:00
ast_channel_unlock ( chan ) ;
}
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 ;
2008-12-12 16:55:15 +00:00
/* We have to pass AST_DEVICE_UNKNOWN here because it is entirely possible that the channel driver
* for this channel is using the callback method for device state . If we pass in an actual state here
* we override what they are saying the state is and things go amuck . */
ast_devstate_changed_literal ( AST_DEVICE_UNKNOWN , name ) ;
2006-11-07 21:47:49 +00:00
/* setstate used to conditionally report Newchannel; this is no more */
2009-11-13 20:42:03 +00:00
ast_manager_event ( chan , EVENT_FLAG_CALL , " Newstate " ,
2010-07-14 15:48:36 +00:00
" Channel: %s \r \n "
" ChannelState: %d \r \n "
" ChannelStateDesc: %s \r \n "
" CallerIDNum: %s \r \n "
" CallerIDName: %s \r \n "
2011-05-25 17:06:38 +00:00
" ConnectedLineNum: %s \r \n "
" ConnectedLineName: %s \r \n "
2010-07-14 15:48:36 +00:00
" Uniqueid: %s \r \n " ,
chan - > name , chan - > _state , ast_state2str ( chan - > _state ) ,
S_COR ( chan - > caller . id . number . valid , chan - > caller . id . number . str , " " ) ,
S_COR ( chan - > caller . id . name . valid , chan - > caller . id . name . str , " " ) ,
2011-05-25 17:06:38 +00:00
S_COR ( chan - > connected . id . number . valid , chan - > connected . id . number . str , " " ) ,
S_COR ( chan - > connected . id . name . valid , chan - > connected . id . name . str , " " ) ,
2010-07-14 15:48:36 +00:00
chan - > uniqueid ) ;
2005-09-01 22:28:33 +00:00
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 ,
2009-04-08 21:00:39 +00:00
struct ast_channel * * rc )
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 ;
2009-11-04 14:05:12 +00:00
format_t o0nativeformats ;
format_t 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 ) ;
2009-05-01 18:09:23 +00:00
if ( config - > feature_timer > 0 & & ast_tvzero ( config - > nexteventts ) ) {
/* nexteventts is not set when the bridge is not scheduled to
* break , so calculate when the bridge should possibly break
* if a partial feature match timed out */
config - > nexteventts = ast_tvadd ( ast_tvnow ( ) , ast_samp2tv ( config - > feature_timer , 1000 ) ) ;
}
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 ;
}
2009-04-08 21:00:39 +00:00
if ( config - > nexteventts . tv_sec ) {
to = ast_tvdiff_ms ( config - > nexteventts , ast_tvnow ( ) ) ;
2006-01-25 09:50:28 +00:00
if ( to < = 0 ) {
2009-04-08 21:00:39 +00:00
if ( config - > timelimit & & ! config - > feature_timer & & ! ast_test_flag ( config , AST_FEATURE_WARNING_ACTIVE ) ) {
2006-11-15 21:36:13 +00:00
res = AST_BRIDGE_RETRY ;
2009-02-17 22:08:00 +00:00
/* generic bridge ending to play warning */
ast_set_flag ( config , AST_FEATURE_WARNING_ACTIVE ) ;
2009-04-08 21:00:39 +00:00
} else if ( config - > feature_timer ) {
/* feature timer expired - make sure we do not play warning */
ast_clear_flag ( config , AST_FEATURE_WARNING_ACTIVE ) ;
res = AST_BRIDGE_RETRY ;
2009-02-17 22:08:00 +00:00
} else {
2006-11-15 21:36:13 +00:00
res = AST_BRIDGE_COMPLETE ;
2009-02-17 22:08:00 +00:00
}
2006-01-25 09:50:28 +00:00
break ;
}
2009-05-01 18:09:23 +00:00
} else {
/* If a feature has been started and the bridge is configured to
* to not break , leave the channel bridge when the feature timer
* time has elapsed so the DTMF will be sent to the other side .
*/
if ( ! ast_tvzero ( config - > nexteventts ) ) {
int diff = ast_tvdiff_ms ( config - > nexteventts , ast_tvnow ( ) ) ;
if ( diff < = 0 ) {
res = AST_BRIDGE_RETRY ;
break ;
}
}
2006-01-25 09:50:28 +00:00
to = - 1 ;
2009-05-01 18:09:23 +00:00
}
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 ) ;
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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 ) ;
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if ( ( c0 - > _softhangup | c1 - > _softhangup ) & AST_SOFTHANGUP_UNBRIDGE ) { /* Bit operators are intentional. */
if ( c0 - > _softhangup & AST_SOFTHANGUP_UNBRIDGE ) {
Merged revisions 303548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
Merged revisions 303546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
Fix channel redirect out of MeetMe() and other issues with channel softhangup.
Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
working properly. This issue includes a patch that resolves the issue by
removing a call to ast_check_hangup() from app_meetme.c. I left that in my
patch, as it doesn't need to be there. However, the rest of the patch fixes
this problem with or without the change to app_meetme.
The key difference between what happens before and after this patch is the
effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(),
ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme
sees this which causes it to exit as intended. Checking ast_check_hangup()
caused app_meetme to exit earlier in the process, and the target of the
redirect saw the condition where ast_read() returned NULL.
Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
solve the issue if another application did the same thing. There are also
other edge cases where if an application finishes at the same time that a
redirect happens, the target of the redirect will think that the channel hung
up. So, I made some changes in pbx.c to resolve it at a deeper level. There
are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
abort the hangup process. My patch extends this to remove the END_OF_Q frame
from the channel's read queue, making the "abort hangup" more complete. This
same technique was used in every place where a softhangup flag was cleared.
(closes issue #18585)
Reported by: oej
Tested by: oej, wedhorn, russell
Review: https://reviewboard.asterisk.org/r/1082/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-24 20:51:37 +00:00
ast_channel_clear_softhangup ( c0 , AST_SOFTHANGUP_UNBRIDGE ) ;
2010-11-22 19:36:10 +00:00
}
if ( c1 - > _softhangup & AST_SOFTHANGUP_UNBRIDGE ) {
Merged revisions 303548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
Merged revisions 303546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
Fix channel redirect out of MeetMe() and other issues with channel softhangup.
Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
working properly. This issue includes a patch that resolves the issue by
removing a call to ast_check_hangup() from app_meetme.c. I left that in my
patch, as it doesn't need to be there. However, the rest of the patch fixes
this problem with or without the change to app_meetme.
The key difference between what happens before and after this patch is the
effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(),
ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme
sees this which causes it to exit as intended. Checking ast_check_hangup()
caused app_meetme to exit earlier in the process, and the target of the
redirect saw the condition where ast_read() returned NULL.
Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
solve the issue if another application did the same thing. There are also
other edge cases where if an application finishes at the same time that a
redirect happens, the target of the redirect will think that the channel hung
up. So, I made some changes in pbx.c to resolve it at a deeper level. There
are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
abort the hangup process. My patch extends this to remove the END_OF_Q frame
from the channel's read queue, making the "abort hangup" more complete. This
same technique was used in every place where a softhangup flag was cleared.
(closes issue #18585)
Reported by: oej
Tested by: oej, wedhorn, russell
Review: https://reviewboard.asterisk.org/r/1082/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-24 20:51:37 +00:00
ast_channel_clear_softhangup ( c1 , AST_SOFTHANGUP_UNBRIDGE ) ;
2010-11-22 19:36:10 +00:00
}
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c0 - > _bridge = c1 ;
c1 - > _bridge = c0 ;
}
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continue ;
}
f = ast_read ( who ) ;
if ( ! f ) {
* fo = NULL ;
* rc = who ;
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ast_debug ( 1 , " Didn't get a frame from channel: %s \n " , who - > name ) ;
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break ;
}
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other = ( who = = c0 ) ? c1 : c0 ; /* the 'other' channel */
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/* 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
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if ( ( f - > frametype = = AST_FRAME_CONTROL ) & & ! ( config - > flags & AST_BRIDGE_IGNORE_SIGS ) ) {
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int bridge_exit = 0 ;
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switch ( f - > subclass . integer ) {
Generic Advice of Charge.
Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages
Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup
AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.
SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option.
IAX AOC Support
- Natively supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type
DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
example usage:
;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))
Review: https://reviewboard.asterisk.org/r/552/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02 18:10:15 +00:00
case AST_CONTROL_AOC :
ast_indicate_data ( other , f - > subclass . integer , f - > data . ptr , f - > datalen ) ;
break ;
2009-06-01 20:57:31 +00:00
case AST_CONTROL_REDIRECTING :
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
if ( ast_channel_redirecting_macro ( who , other , f , other = = c0 , 1 ) ) {
ast_indicate_data ( other , f - > subclass . integer , f - > data . ptr , f - > datalen ) ;
}
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break ;
case AST_CONTROL_CONNECTED_LINE :
if ( ast_channel_connected_line_macro ( who , other , f , other = = c0 , 1 ) ) {
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ast_indicate_data ( other , f - > subclass . integer , f - > data . ptr , f - > datalen ) ;
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}
break ;
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case AST_CONTROL_HOLD :
case AST_CONTROL_UNHOLD :
case AST_CONTROL_VIDUPDATE :
2009-04-03 22:41:46 +00:00
case AST_CONTROL_SRCUPDATE :
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case AST_CONTROL_SRCCHANGE :
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case AST_CONTROL_T38_PARAMETERS :
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ast_indicate_data ( other , f - > subclass . integer , f - > data . ptr , f - > datalen ) ;
2008-08-06 00:30:53 +00:00
if ( jb_in_use ) {
ast_jb_empty_and_reset ( c0 , c1 ) ;
}
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break ;
default :
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* fo = f ;
* rc = who ;
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bridge_exit = 1 ;
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ast_debug ( 1 , " Got a FRAME_CONTROL (%d) frame on channel %s \n " , f - > subclass . integer , who - > name ) ;
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break ;
}
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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 ) | |
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( f - > frametype = = AST_FRAME_DTMF_BEGIN ) | |
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( 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 ;
2009-04-08 21:00:39 +00:00
if ( monitored_source & &
( f - > frametype = = AST_FRAME_DTMF_END | |
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f - > frametype = = AST_FRAME_DTMF_BEGIN ) ) {
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* 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 ) ;
2009-02-17 22:08:00 +00:00
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
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/* 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 )
{
2009-11-13 20:42:03 +00:00
struct ast_channel * chans [ 2 ] = { c0 , c1 } ;
ast_manager_event_multichan ( EVENT_FLAG_CALL , " Bridge " , 2 , chans ,
2010-07-14 15:48:36 +00:00
" 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_COR ( c0 - > caller . id . number . valid , c0 - > caller . id . number . str , " " ) ,
S_COR ( c1 - > caller . id . number . valid , c1 - > caller . id . number . str , " " ) ) ;
2007-12-06 15:29:05 +00:00
}
2009-03-11 22:14:55 +00:00
static void update_bridge_vars ( struct ast_channel * c0 , struct ast_channel * c1 )
2009-03-11 21:49:29 +00:00
{
const char * c0_name ;
const char * c1_name ;
2009-03-11 22:14:55 +00:00
const char * c0_pvtid = NULL ;
const char * c1_pvtid = NULL ;
2009-03-11 21:49:29 +00:00
ast_channel_lock ( c1 ) ;
c1_name = ast_strdupa ( c1 - > name ) ;
2009-03-11 22:14:55 +00:00
if ( c1 - > tech - > get_pvt_uniqueid ) {
c1_pvtid = ast_strdupa ( c1 - > tech - > get_pvt_uniqueid ( c1 ) ) ;
}
2009-03-11 21:49:29 +00:00
ast_channel_unlock ( c1 ) ;
ast_channel_lock ( c0 ) ;
if ( ! ast_strlen_zero ( pbx_builtin_getvar_helper ( c0 , " BRIDGEPEER " ) ) ) {
pbx_builtin_setvar_helper ( c0 , " BRIDGEPEER " , c1_name ) ;
}
2009-03-11 22:14:55 +00:00
if ( c1_pvtid ) {
pbx_builtin_setvar_helper ( c0 , " BRIDGEPVTCALLID " , c1_pvtid ) ;
}
2009-03-11 21:49:29 +00:00
c0_name = ast_strdupa ( c0 - > name ) ;
2009-03-11 22:14:55 +00:00
if ( c0 - > tech - > get_pvt_uniqueid ) {
c0_pvtid = ast_strdupa ( c0 - > tech - > get_pvt_uniqueid ( c0 ) ) ;
}
2009-03-11 21:49:29 +00:00
ast_channel_unlock ( c0 ) ;
ast_channel_lock ( c1 ) ;
if ( ! ast_strlen_zero ( pbx_builtin_getvar_helper ( c1 , " BRIDGEPEER " ) ) ) {
pbx_builtin_setvar_helper ( c1 , " BRIDGEPEER " , c0_name ) ;
}
2009-03-11 22:14:55 +00:00
if ( c0_pvtid ) {
pbx_builtin_setvar_helper ( c1 , " BRIDGEPVTCALLID " , c0_pvtid ) ;
}
2009-03-11 21:49:29 +00:00
ast_channel_unlock ( c1 ) ;
}
2009-03-11 22:25:57 +00:00
static void bridge_play_sounds ( struct ast_channel * c0 , struct ast_channel * c1 )
{
const char * s , * sound ;
/* See if we need to play an audio file to any side of the bridge */
ast_channel_lock ( c0 ) ;
if ( ( s = pbx_builtin_getvar_helper ( c0 , " BRIDGE_PLAY_SOUND " ) ) ) {
sound = ast_strdupa ( s ) ;
ast_channel_unlock ( c0 ) ;
bridge_playfile ( c0 , c1 , sound , 0 ) ;
pbx_builtin_setvar_helper ( c0 , " BRIDGE_PLAY_SOUND " , NULL ) ;
} else {
ast_channel_unlock ( c0 ) ;
}
ast_channel_lock ( c1 ) ;
if ( ( s = pbx_builtin_getvar_helper ( c1 , " BRIDGE_PLAY_SOUND " ) ) ) {
sound = ast_strdupa ( s ) ;
ast_channel_unlock ( c1 ) ;
bridge_playfile ( c1 , c0 , sound , 0 ) ;
pbx_builtin_setvar_helper ( c1 , " BRIDGE_PLAY_SOUND " , NULL ) ;
} else {
ast_channel_unlock ( c1 ) ;
}
}
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 ,
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struct ast_bridge_config * config , struct ast_frame * * fo , struct ast_channel * * rc )
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{
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struct ast_channel * chans [ 2 ] = { c0 , c1 } ;
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enum ast_bridge_result res = AST_BRIDGE_COMPLETE ;
2009-11-04 14:05:12 +00:00
format_t o0nativeformats ;
format_t o1nativeformats ;
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long time_left_ms = 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
2011-05-20 16:43:02 +00:00
* fo = NULL ;
2004-10-23 12:19:47 +00:00
if ( c0 - > _bridge ) {
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ast_log ( LOG_WARNING , " %s is already in a bridge with %s \n " ,
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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 ) )
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return - 1 ;
2011-09-12 15:54:41 +00:00
caller_warning = ast_test_flag ( & config - > features_caller , AST_FEATURE_PLAY_WARNING ) ;
callee_warning = ast_test_flag ( & config - > features_callee , AST_FEATURE_PLAY_WARNING ) ;
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if ( ast_tvzero ( config - > start_time ) ) {
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config - > start_time = ast_tvnow ( ) ;
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if ( config - > start_sound ) {
if ( caller_warning ) {
bridge_playfile ( c0 , c1 , config - > start_sound , config - > timelimit / 1000 ) ;
}
if ( callee_warning ) {
bridge_playfile ( c1 , c0 , config - > start_sound , config - > timelimit / 1000 ) ;
}
}
2005-08-09 01:59:59 +00:00
}
2001-05-07 00:43:32 +00:00
/* Keep track of bridge */
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c0 - > _bridge = c1 ;
c1 - > _bridge = c0 ;
2006-04-21 10:05:45 +00:00
2009-06-26 15:28:53 +00:00
ast_set_owners_and_peers ( c0 , c1 ) ;
2006-04-06 15:40:12 +00:00
2004-09-04 01:06:42 +00:00
o0nativeformats = c0 - > nativeformats ;
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o1nativeformats = c1 - > nativeformats ;
2009-03-03 00:01:51 +00:00
if ( config - > feature_timer & & ! ast_tvzero ( config - > nexteventts ) ) {
2009-04-08 21:00:39 +00:00
config - > nexteventts = ast_tvadd ( config - > feature_start_time , ast_samp2tv ( config - > feature_timer , 1000 ) ) ;
} else if ( config - > timelimit ) {
time_left_ms = config - > timelimit - ast_tvdiff_ms ( ast_tvnow ( ) , config - > start_time ) ;
2009-02-17 22:08:00 +00:00
config - > nexteventts = ast_tvadd ( config - > start_time , ast_samp2tv ( config - > timelimit , 1000 ) ) ;
2009-04-08 21:00:39 +00:00
if ( ( caller_warning | | callee_warning ) & & config - > play_warning ) {
long next_warn = config - > play_warning ;
2011-06-13 19:51:52 +00:00
if ( time_left_ms < config - > play_warning & & config - > warning_freq > 0 ) {
2009-04-08 21:00:39 +00:00
/* At least one warning was played, which means we are returning after feature */
long warns_passed = ( config - > play_warning - time_left_ms ) / config - > warning_freq ;
/* It is 'warns_passed * warning_freq' NOT '(warns_passed + 1) * warning_freq',
because nexteventts will be updated once again in the ' if ( ! to ) ' block */
next_warn = config - > play_warning - warns_passed * config - > warning_freq ;
}
config - > nexteventts = ast_tvsub ( config - > nexteventts , ast_samp2tv ( next_warn , 1000 ) ) ;
}
} else {
config - > nexteventts . tv_sec = 0 ;
config - > nexteventts . tv_usec = 0 ;
2005-10-13 18:27:54 +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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19 18:06:03 +00:00
if ( ! c0 - > tech - > send_digit_begin )
ast_set_flag ( c1 , AST_FLAG_END_DTMF_ONLY ) ;
if ( ! c1 - > tech - > send_digit_begin )
ast_set_flag ( c0 , AST_FLAG_END_DTMF_ONLY ) ;
2007-12-06 15:29:05 +00:00
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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 */
2010-12-07 22:59:30 +00:00
ast_indicate ( c0 , AST_CONTROL_SRCUPDATE ) ;
ast_indicate ( c1 , AST_CONTROL_SRCUPDATE ) ;
2008-03-05 22:43:22 +00:00
2001-03-12 03:04:51 +00:00
for ( /* ever */ ; ; ) {
2006-08-05 05:26:29 +00:00
struct timeval now = { 0 , } ;
2006-08-01 23:09:28 +00:00
int to ;
2005-10-13 18:27:54 +00:00
to = - 1 ;
2006-08-01 23:09:28 +00:00
2009-02-17 22:08:00 +00:00
if ( ! ast_tvzero ( config - > nexteventts ) ) {
2005-10-13 18:27:54 +00:00
now = ast_tvnow ( ) ;
2009-02-17 22:08:00 +00:00
to = ast_tvdiff_ms ( config - > 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: https://origsvn.digium.com/svn/asterisk/trunk@43780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@43780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 ;
res = 0 ;
2004-04-26 23:22:34 +00:00
break ;
}
2009-03-11 21:49:29 +00:00
2005-10-13 18:27:54 +00:00
if ( ! to ) {
2009-02-17 22:08:00 +00:00
if ( time_left_ms > = 5000 & & config - > warning_sound & & config - > play_warning & & ast_test_flag ( config , AST_FEATURE_WARNING_ACTIVE ) ) {
2006-04-16 16:04:24 +00:00
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
}
2009-04-08 21:00:39 +00:00
if ( config - > warning_freq & & ( time_left_ms > ( config - > warning_freq + 5000 ) ) ) {
2009-02-17 22:08:00 +00:00
config - > nexteventts = ast_tvadd ( config - > nexteventts , ast_samp2tv ( config - > warning_freq , 1000 ) ) ;
2009-04-08 21:00:39 +00:00
} else {
2009-02-17 22:08:00 +00:00
config - > nexteventts = ast_tvadd ( config - > start_time , ast_samp2tv ( config - > timelimit , 1000 ) ) ;
2009-04-08 21:00:39 +00:00
}
2004-04-26 23:22:34 +00:00
}
2009-02-17 22:08:00 +00:00
ast_clear_flag ( config , AST_FEATURE_WARNING_ACTIVE ) ;
2004-04-26 23:22:34 +00:00
}
2005-03-23 21:52:31 +00:00
2010-11-22 19:36:10 +00:00
if ( ( c0 - > _softhangup | c1 - > _softhangup ) & AST_SOFTHANGUP_UNBRIDGE ) { /* Bit operators are intentional. */
if ( c0 - > _softhangup & AST_SOFTHANGUP_UNBRIDGE ) {
Merged revisions 303548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
Merged revisions 303546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
Fix channel redirect out of MeetMe() and other issues with channel softhangup.
Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
working properly. This issue includes a patch that resolves the issue by
removing a call to ast_check_hangup() from app_meetme.c. I left that in my
patch, as it doesn't need to be there. However, the rest of the patch fixes
this problem with or without the change to app_meetme.
The key difference between what happens before and after this patch is the
effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(),
ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme
sees this which causes it to exit as intended. Checking ast_check_hangup()
caused app_meetme to exit earlier in the process, and the target of the
redirect saw the condition where ast_read() returned NULL.
Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
solve the issue if another application did the same thing. There are also
other edge cases where if an application finishes at the same time that a
redirect happens, the target of the redirect will think that the channel hung
up. So, I made some changes in pbx.c to resolve it at a deeper level. There
are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
abort the hangup process. My patch extends this to remove the END_OF_Q frame
from the channel's read queue, making the "abort hangup" more complete. This
same technique was used in every place where a softhangup flag was cleared.
(closes issue #18585)
Reported by: oej
Tested by: oej, wedhorn, russell
Review: https://reviewboard.asterisk.org/r/1082/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-24 20:51:37 +00:00
ast_channel_clear_softhangup ( c0 , AST_SOFTHANGUP_UNBRIDGE ) ;
2010-11-22 19:36:10 +00:00
}
if ( c1 - > _softhangup & AST_SOFTHANGUP_UNBRIDGE ) {
Merged revisions 303548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
Merged revisions 303546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
Fix channel redirect out of MeetMe() and other issues with channel softhangup.
Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
working properly. This issue includes a patch that resolves the issue by
removing a call to ast_check_hangup() from app_meetme.c. I left that in my
patch, as it doesn't need to be there. However, the rest of the patch fixes
this problem with or without the change to app_meetme.
The key difference between what happens before and after this patch is the
effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(),
ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme
sees this which causes it to exit as intended. Checking ast_check_hangup()
caused app_meetme to exit earlier in the process, and the target of the
redirect saw the condition where ast_read() returned NULL.
Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
solve the issue if another application did the same thing. There are also
other edge cases where if an application finishes at the same time that a
redirect happens, the target of the redirect will think that the channel hung
up. So, I made some changes in pbx.c to resolve it at a deeper level. There
are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
abort the hangup process. My patch extends this to remove the END_OF_Q frame
from the channel's read queue, making the "abort hangup" more complete. This
same technique was used in every place where a softhangup flag was cleared.
(closes issue #18585)
Reported by: oej
Tested by: oej, wedhorn, russell
Review: https://reviewboard.asterisk.org/r/1082/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-24 20:51:37 +00:00
ast_channel_clear_softhangup ( c1 , AST_SOFTHANGUP_UNBRIDGE ) ;
2010-11-22 19:36:10 +00:00
}
2005-03-23 21:52:31 +00:00
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 ;
}
2009-03-11 21:49:29 +00:00
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 ;
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 ;
}
2009-03-11 21:49:29 +00:00
2009-03-11 22:14:55 +00:00
update_bridge_vars ( c0 , c1 ) ;
2008-03-25 15:18:41 +00:00
2009-03-11 22:25:57 +00:00
bridge_play_sounds ( c0 , c1 ) ;
2009-03-11 21:49:29 +00:00
2005-08-09 01:59:59 +00:00
if ( c0 - > tech - > bridge & &
2010-03-23 21:17:23 +00:00
/* if < 1 ms remains use generic bridging for accurate timing */
( ! config - > timelimit | | to > 1000 | | to = = 0 ) & &
2005-08-09 01:59:59 +00:00
( c0 - > tech - > bridge = = c1 - > tech - > bridge ) & &
2010-02-18 18:31:44 +00:00
! c0 - > monitor & & ! c1 - > monitor & &
2010-03-23 21:17:23 +00:00
! c0 - > audiohooks & & ! c1 - > audiohooks & &
2011-06-29 16:18:39 +00:00
ast_framehook_list_is_empty ( c0 - > framehooks ) & & ast_framehook_list_is_empty ( c1 - > framehooks ) & &
2007-10-08 15:05:05 +00:00
! c0 - > masq & & ! c0 - > masqr & & ! c1 - > masq & & ! c1 - > masqr ) {
2010-03-23 21:17:23 +00:00
int timeoutms = to - 1000 > 0 ? to - 1000 : to ;
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 ) ;
2010-03-23 21:17:23 +00:00
if ( ( res = c0 - > tech - > bridge ( c0 , c1 , config - > flags , fo , rc , timeoutms ) ) = = AST_BRIDGE_COMPLETE ) {
2009-11-13 20:42:03 +00:00
ast_manager_event_multichan ( EVENT_FLAG_CALL , " Unlink " , 2 , chans ,
2010-07-14 15:48:36 +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 ,
S_COR ( c0 - > caller . id . number . valid , c0 - > caller . id . number . str , " <unknown> " ) ,
S_COR ( c1 - > caller . id . number . valid , c1 - > caller . id . number . str , " <unknown> " ) ) ;
2009-06-26 15:28:53 +00:00
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
2010-11-22 19:36:10 +00:00
if ( ( c0 - > _softhangup | c1 - > _softhangup ) & AST_SOFTHANGUP_UNBRIDGE ) { /* Bit operators are intentional. */
2005-03-23 21:52:31 +00:00
continue ;
2010-11-22 19:36:10 +00:00
}
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 :
2009-04-07 23:50:56 +00:00
if ( config - > play_warning ) {
ast_set_flag ( config , AST_FEATURE_WARNING_ACTIVE ) ;
}
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 :
break ;
}
2001-09-19 14:40:16 +00:00
}
2009-03-11 21:49:29 +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 ) ;
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return AST_BRIDGE_FAILED ;
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}
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o0nativeformats = c0 - > nativeformats ;
o1nativeformats = c1 - > nativeformats ;
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}
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update_bridge_vars ( c0 , c1 ) ;
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res = ast_generic_bridge ( c0 , c1 , config , fo , rc ) ;
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if ( res ! = AST_BRIDGE_RETRY ) {
break ;
} else if ( config - > feature_timer ) {
/* feature timer expired but has not been updated, sending to ast_bridge_call to do so */
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break ;
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}
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}
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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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 ) ;
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/* 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 ) ;
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c0 - > _bridge = NULL ;
c1 - > _bridge = NULL ;
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ast_manager_event_multichan ( EVENT_FLAG_CALL , " Unlink " , 2 , chans ,
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" 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 ,
S_COR ( c0 - > caller . id . number . valid , c0 - > caller . id . number . str , " <unknown> " ) ,
S_COR ( c1 - > caller . id . number . valid , c1 - > caller . id . number . str , " <unknown> " ) ) ;
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ast_debug ( 1 , " Bridge stops bridging channels %s and %s \n " , c0 - > name , c1 - > name ) ;
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return res ;
}
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/*! \brief Sets an option on a channel */
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int ast_channel_setoption ( struct ast_channel * chan , int option , void * data , int datalen , int block )
{
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int res ;
ast_channel_lock ( chan ) ;
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if ( ! chan - > tech - > setoption ) {
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errno = ENOSYS ;
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ast_channel_unlock ( chan ) ;
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return - 1 ;
}
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if ( block )
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ast_log ( LOG_ERROR , " XXX Blocking not implemented yet XXX \n " ) ;
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res = chan - > tech - > setoption ( chan , option , data , datalen ) ;
ast_channel_unlock ( chan ) ;
return res ;
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}
int ast_channel_queryoption ( struct ast_channel * chan , int option , void * data , int * datalen , int block )
{
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int res ;
ast_channel_lock ( chan ) ;
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if ( ! chan - > tech - > queryoption ) {
errno = ENOSYS ;
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ast_channel_unlock ( chan ) ;
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return - 1 ;
}
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if ( block )
ast_log ( LOG_ERROR , " XXX Blocking not implemented yet XXX \n " ) ;
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res = chan - > tech - > queryoption ( chan , option , data , datalen ) ;
ast_channel_unlock ( chan ) ;
return res ;
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}
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struct tonepair_def {
int freq1 ;
int freq2 ;
int duration ;
int vol ;
} ;
struct tonepair_state {
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int fac1 ;
int fac2 ;
int v1_1 ;
int v2_1 ;
int v3_1 ;
int v1_2 ;
int v2_2 ;
int v3_2 ;
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format_t origwfmt ;
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int pos ;
int duration ;
int modulate ;
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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 ;
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if ( chan )
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ast_set_write_format ( chan , ts - > origwfmt ) ;
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ast_free ( ts ) ;
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}
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static void * tonepair_alloc ( struct ast_channel * chan , void * params )
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{
struct tonepair_state * ts ;
struct tonepair_def * td = params ;
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if ( ! ( ts = ast_calloc ( 1 , sizeof ( * ts ) ) ) )
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return NULL ;
ts - > origwfmt = chan - > writeformat ;
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if ( ast_set_write_format ( chan , AST_FORMAT_SLINEAR ) ) {
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ast_log ( LOG_WARNING , " Unable to set '%s' to signed linear format (write) \n " , chan - > name ) ;
tonepair_release ( NULL , ts ) ;
ts = NULL ;
} else {
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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 ;
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ts - > duration = td - > duration ;
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ts - > modulate = 0 ;
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}
/* Let interrupts interrupt :) */
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ast_set_flag ( chan , AST_FLAG_WRITE_INT ) ;
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return ts ;
}
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static int tonepair_generator ( struct ast_channel * chan , void * data , int len , int samples )
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{
struct tonepair_state * ts = data ;
int x ;
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/* we need to prepare a frame with 16 * timelen samples as we're
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* generating SLIN audio
*/
len = samples * 2 ;
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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 ) ) ;
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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 ;
}
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ts - > f . frametype = AST_FRAME_VOICE ;
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ts - > f . subclass . codec = AST_FORMAT_SLINEAR ;
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ts - > f . datalen = len ;
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ts - > f . samples = samples ;
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ts - > f . offset = AST_FRIENDLY_OFFSET ;
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ts - > f . data . ptr = ts - > data ;
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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 , } ;
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d . freq1 = freq1 ;
d . freq2 = freq2 ;
d . duration = duration ;
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d . vol = ( vol < 1 ) ? 8192 : vol ; /* force invalid to 8192 */
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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 ;
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if ( ( res = ast_tonepair_start ( chan , freq1 , freq2 , duration , vol ) ) )
return res ;
/* Give us some wiggle room */
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while ( chan - > generatordata & & ast_waitfor ( chan , 100 ) > = 0 ) {
struct ast_frame * f = ast_read ( chan ) ;
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if ( f )
ast_frfree ( f ) ;
else
return - 1 ;
}
return 0 ;
}
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2006-11-16 08:18:41 +00:00
ast_group_t ast_get_group ( const char * s )
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{
char * piece ;
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char * c ;
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int start = 0 , finish = 0 , x ;
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ast_group_t group = 0 ;
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2007-12-27 21:41:22 +00:00
if ( ast_strlen_zero ( s ) )
return 0 ;
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c = ast_strdupa ( s ) ;
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2005-12-24 05:18:40 +00:00
while ( ( piece = strsep ( & c , " , " ) ) ) {
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if ( sscanf ( piece , " %30d-%30d " , & start , & finish ) = = 2 ) {
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/* Range */
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} else if ( sscanf ( piece , " %30d " , & start ) ) {
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/* Just one */
finish = start ;
} else {
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ast_log ( LOG_ERROR , " Syntax error parsing group configuration '%s' at '%s'. Ignoring. \n " , s , piece ) ;
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continue ;
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}
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for ( x = start ; x < = finish ; x + + ) {
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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
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group | = ( ( ast_group_t ) 1 < < x ) ;
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}
}
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 ;
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static void ( * ast_moh_stop_ptr ) ( struct ast_channel * ) = NULL ;
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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 * ) ,
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void ( * stop_ptr ) ( struct ast_channel * ) ,
void ( * cleanup_ptr ) ( struct ast_channel * ) )
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{
ast_moh_start_ptr = start_ptr ;
ast_moh_stop_ptr = stop_ptr ;
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ast_moh_cleanup_ptr = cleanup_ptr ;
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}
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void ast_uninstall_music_functions ( void )
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{
ast_moh_start_ptr = NULL ;
ast_moh_stop_ptr = NULL ;
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ast_moh_cleanup_ptr = NULL ;
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}
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/*! \brief Turn on music on hold on a given channel */
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int ast_moh_start ( struct ast_channel * chan , const char * mclass , const char * interpclass )
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{
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if ( ast_moh_start_ptr )
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return ast_moh_start_ptr ( chan , mclass , interpclass ) ;
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ast_verb ( 3 , " Music class %s requested but no musiconhold loaded. \n " , mclass ? mclass : ( interpclass ? interpclass : " default " ) ) ;
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return 0 ;
}
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/*! \brief Turn off music on hold on a given channel */
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void ast_moh_stop ( struct ast_channel * chan )
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{
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if ( ast_moh_stop_ptr )
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ast_moh_stop_ptr ( chan ) ;
}
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2006-04-06 15:40:12 +00:00
void ast_moh_cleanup ( struct ast_channel * chan )
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{
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if ( ast_moh_cleanup_ptr )
ast_moh_cleanup_ptr ( chan ) ;
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}
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
static int ast_channel_hash_cb ( const void * obj , const int flags )
{
const struct ast_channel * chan = obj ;
/* If the name isn't set, return 0 so that the ao2_find() search will
* start in the first bucket . */
if ( ast_strlen_zero ( chan - > name ) ) {
return 0 ;
}
return ast_str_case_hash ( chan - > name ) ;
}
Fix transcode_via_sln option with SIP calls and improve PLC usage.
From reviewboard:
The problem here is a bit complex, so try to bear with me...
It was noticed by a Digium customer that generic PLC (as configured in
codecs.conf) did not appear to actually be having any sort of benefit when
packet loss was introduced on an RTP stream. I reproduced this issue myself
by streaming a file across an RTP stream and dropping approx. 5% of the
RTP packets. I saw no real difference between when PLC was enabled or disabled
when using wireshark to analyze the RTP streams.
After analyzing what was going on, it became clear that one of the problems
faced was that when running my tests, the translation paths were being set
up in such a way that PLC could not possibly work as expected. To illustrate,
if packets are lost on channel A's read stream, then we expect that PLC will
be applied to channel B's write stream. The problem is that generic PLC can
only be done when there is a translation path that moves from some codec to
SLINEAR. When I would run my tests, I found that every single time, read
and write translation paths would be set up on channel A instead of channel
B. There appeared to be no real way to predict which channel the translation
paths would be set up on.
This is where Kevin swooped in to let me know about the transcode_via_sln
option in asterisk.conf. It is supposed to work by placing a read translation
path on both channels from the channel's rawreadformat to SLINEAR. It also
will place a write translation path on both channels from SLINEAR to the
channel's rawwriteformat. Using this option allows one to predictably set up
translation paths on all channels. There are two problems with this, though.
First and foremost, the transcode_via_sln option did not appear to be working
properly when I was placing a SIP call between two endpoints which did not
share any common formats. Second, even if this option were to work, for PLC
to be applied, there had to be a write translation path that would go from
some format to SLINEAR. It would not work properly if the starting format
of translation was SLINEAR.
The one-line change presented in this review request in chan_sip.c fixed the
first issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the format passed to
sip_request_call. This is nativeformats of the inbound channel. Because of this,
when ast_channel_make_compatible was called by app_dial, both channels already
had compatibly read and write formats. Thus, no translation path was set up at
the time. My change is to set the jointcapability of the sip_pvt created during
sip_request_call to the intersection of the inbound channel's nativeformats and
the configured peer capability that we determined during the earlier call to
create_addr. Doing this got the translation paths set up as expected when using
transcode_via_sln.
The changes presented in channel.c fixed the second issue for me. First and
foremost, when Asterisk is started, we'll read codecs.conf to see the value of
the genericplc option. If this option is set, and ast_write is called for a
frame with no data, then we will attempt to fill in the missing samples for
the frame. The implementation uses a channel datastore for maintaining the
PLC state and for creating a buffer to store PLC samples in. Even when we
receive a frame with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a basis for when
it comes time to actually do a PLC fill-in.
So, reviewers, now I ask for your help. First off, there's the one line change
in chan_sip that I have put in. Is it right? By my logic it seems correct, but
I'm sure someone can tell me why it is not going to work. This is probably the
change I'm least concerned about, though. What concerns me much more is the
set of changes in channel.c. First off, am I even doing it right? When I run
tests, I can clearly see that when PLC is activated, I see a significant increase
in RTP traffic where I would expect it to be. However, in my humble opinion, the
audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to
me than when no PLC is used at all. I need someone to review the logic I have used
to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic
is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm
sure someone can point out somewhere where I've done something incorrectly.
As I was writing this review request up, I decided to give the code a test run under
valgrind, and I find that for some reason, calls to plc_rx are causing some invalid
reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around
a bit to see why that is the case. If it's obvious to someone reviewing, speak up!
Finally, I have one other proposal that is not reflected in my code review. Since
without transcode_via_sln set, one cannot predict or control where a translation
path will be up, it seems to me that the current practice of using PLC only when
transcoding to SLINEAR is not useful. I recommend that once it has been determined
that the method used in this code review is correct and works as expected, then
the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 21:29:08 +00:00
int ast_plc_reload ( void )
{
struct ast_variable * var ;
struct ast_flags config_flags = { 0 } ;
struct ast_config * cfg = ast_config_load ( " codecs.conf " , config_flags ) ;
if ( cfg = = CONFIG_STATUS_FILEMISSING | | cfg = = CONFIG_STATUS_FILEUNCHANGED | | cfg = = CONFIG_STATUS_FILEINVALID )
return 0 ;
for ( var = ast_variable_browse ( cfg , " plc " ) ; var ; var = var - > next ) {
if ( ! strcasecmp ( var - > name , " genericplc " ) ) {
ast_set2_flag ( & ast_options , ast_true ( var - > value ) , AST_OPT_FLAG_GENERIC_PLC ) ;
}
}
ast_config_destroy ( cfg ) ;
return 0 ;
}
2010-04-22 18:07:02 +00:00
/*!
* \ internal
* \ brief Implements the channels provider .
*/
static int data_channels_provider_handler ( const struct ast_data_search * search ,
struct ast_data * root )
{
2010-07-08 14:48:42 +00:00
struct ast_channel * c ;
2010-04-22 18:07:02 +00:00
struct ast_channel_iterator * iter = NULL ;
2010-07-08 14:48:42 +00:00
struct ast_data * data_channel ;
2010-04-22 18:07:02 +00:00
for ( iter = ast_channel_iterator_all_new ( ) ;
iter & & ( c = ast_channel_iterator_next ( iter ) ) ; ast_channel_unref ( c ) ) {
ast_channel_lock ( c ) ;
data_channel = ast_data_add_node ( root , " channel " ) ;
if ( ! data_channel ) {
ast_channel_unlock ( c ) ;
continue ;
}
2010-07-08 14:48:42 +00:00
if ( ast_channel_data_add_structure ( data_channel , c , 1 ) < 0 ) {
ast_log ( LOG_ERROR , " Unable to add channel structure for channel: %s \n " , c - > name ) ;
2010-04-22 18:07:02 +00:00
}
ast_channel_unlock ( c ) ;
2010-07-08 14:48:42 +00:00
if ( ! ast_data_search_match ( search , data_channel ) ) {
ast_data_remove_node ( root , data_channel ) ;
}
2010-04-22 18:07:02 +00:00
}
if ( iter ) {
ast_channel_iterator_destroy ( iter ) ;
}
return 0 ;
}
2010-07-08 14:48:42 +00:00
/*!
* \ internal
* \ brief Implements the channeltypes provider .
*/
static int data_channeltypes_provider_handler ( const struct ast_data_search * search ,
struct ast_data * data_root )
{
struct chanlist * cl ;
struct ast_data * data_type ;
AST_RWLIST_RDLOCK ( & backends ) ;
AST_RWLIST_TRAVERSE ( & backends , cl , list ) {
data_type = ast_data_add_node ( data_root , " type " ) ;
if ( ! data_type ) {
continue ;
}
ast_data_add_str ( data_type , " name " , cl - > tech - > type ) ;
ast_data_add_str ( data_type , " description " , cl - > tech - > description ) ;
ast_data_add_bool ( data_type , " devicestate " , cl - > tech - > devicestate ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " indications " , cl - > tech - > indicate ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " transfer " , cl - > tech - > transfer ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " send_digit_begin " , cl - > tech - > send_digit_begin ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " send_digit_end " , cl - > tech - > send_digit_end ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " call " , cl - > tech - > call ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " hangup " , cl - > tech - > hangup ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " answer " , cl - > tech - > answer ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " read " , cl - > tech - > read ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " write " , cl - > tech - > write ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " send_text " , cl - > tech - > send_text ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " send_image " , cl - > tech - > send_image ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " send_html " , cl - > tech - > send_html ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " exception " , cl - > tech - > exception ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " bridge " , cl - > tech - > bridge ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " early_bridge " , cl - > tech - > early_bridge ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " fixup " , cl - > tech - > fixup ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " setoption " , cl - > tech - > setoption ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " queryoption " , cl - > tech - > queryoption ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " write_video " , cl - > tech - > write_video ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " write_text " , cl - > tech - > write_text ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " bridged_channel " , cl - > tech - > bridged_channel ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " func_channel_read " , cl - > tech - > func_channel_read ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " func_channel_write " , cl - > tech - > func_channel_write ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " get_base_channel " , cl - > tech - > get_base_channel ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " set_base_channel " , cl - > tech - > set_base_channel ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " get_pvt_uniqueid " , cl - > tech - > get_pvt_uniqueid ? 1 : 0 ) ;
ast_data_add_bool ( data_type , " cc_callback " , cl - > tech - > cc_callback ? 1 : 0 ) ;
ast_data_add_codecs ( data_type , " capabilities " , cl - > tech - > capabilities ) ;
if ( ! ast_data_search_match ( search , data_type ) ) {
ast_data_remove_node ( data_root , data_type ) ;
}
}
AST_RWLIST_UNLOCK ( & backends ) ;
return 0 ;
}
2010-04-22 18:07:02 +00:00
/*!
* \ internal
* \ brief / asterisk / core / channels provider .
*/
static const struct ast_data_handler channels_provider = {
. version = AST_DATA_HANDLER_VERSION ,
. get = data_channels_provider_handler
} ;
2010-07-08 14:48:42 +00:00
/*!
* \ internal
* \ brief / asterisk / core / channeltypes provider .
*/
static const struct ast_data_handler channeltypes_provider = {
. version = AST_DATA_HANDLER_VERSION ,
. get = data_channeltypes_provider_handler
} ;
2010-04-22 18:07:02 +00:00
static const struct ast_data_entry channel_providers [ ] = {
AST_DATA_ENTRY ( " /asterisk/core/channels " , & channels_provider ) ,
2010-07-08 14:48:42 +00:00
AST_DATA_ENTRY ( " /asterisk/core/channeltypes " , & channeltypes_provider ) ,
2010-04-22 18:07:02 +00:00
} ;
2004-12-31 00:04:41 +00:00
void ast_channels_init ( void )
{
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
channels = ao2_container_alloc ( NUM_CHANNEL_BUCKETS ,
ast_channel_hash_cb , ast_channel_cmp_cb ) ;
2008-01-28 04:30:44 +00:00
ast_cli_register_multiple ( cli_channel , ARRAY_LEN ( cli_channel ) ) ;
2010-04-22 18:07:02 +00:00
ast_data_register_multiple_core ( channel_providers , ARRAY_LEN ( channel_providers ) ) ;
Fix transcode_via_sln option with SIP calls and improve PLC usage.
From reviewboard:
The problem here is a bit complex, so try to bear with me...
It was noticed by a Digium customer that generic PLC (as configured in
codecs.conf) did not appear to actually be having any sort of benefit when
packet loss was introduced on an RTP stream. I reproduced this issue myself
by streaming a file across an RTP stream and dropping approx. 5% of the
RTP packets. I saw no real difference between when PLC was enabled or disabled
when using wireshark to analyze the RTP streams.
After analyzing what was going on, it became clear that one of the problems
faced was that when running my tests, the translation paths were being set
up in such a way that PLC could not possibly work as expected. To illustrate,
if packets are lost on channel A's read stream, then we expect that PLC will
be applied to channel B's write stream. The problem is that generic PLC can
only be done when there is a translation path that moves from some codec to
SLINEAR. When I would run my tests, I found that every single time, read
and write translation paths would be set up on channel A instead of channel
B. There appeared to be no real way to predict which channel the translation
paths would be set up on.
This is where Kevin swooped in to let me know about the transcode_via_sln
option in asterisk.conf. It is supposed to work by placing a read translation
path on both channels from the channel's rawreadformat to SLINEAR. It also
will place a write translation path on both channels from SLINEAR to the
channel's rawwriteformat. Using this option allows one to predictably set up
translation paths on all channels. There are two problems with this, though.
First and foremost, the transcode_via_sln option did not appear to be working
properly when I was placing a SIP call between two endpoints which did not
share any common formats. Second, even if this option were to work, for PLC
to be applied, there had to be a write translation path that would go from
some format to SLINEAR. It would not work properly if the starting format
of translation was SLINEAR.
The one-line change presented in this review request in chan_sip.c fixed the
first issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the format passed to
sip_request_call. This is nativeformats of the inbound channel. Because of this,
when ast_channel_make_compatible was called by app_dial, both channels already
had compatibly read and write formats. Thus, no translation path was set up at
the time. My change is to set the jointcapability of the sip_pvt created during
sip_request_call to the intersection of the inbound channel's nativeformats and
the configured peer capability that we determined during the earlier call to
create_addr. Doing this got the translation paths set up as expected when using
transcode_via_sln.
The changes presented in channel.c fixed the second issue for me. First and
foremost, when Asterisk is started, we'll read codecs.conf to see the value of
the genericplc option. If this option is set, and ast_write is called for a
frame with no data, then we will attempt to fill in the missing samples for
the frame. The implementation uses a channel datastore for maintaining the
PLC state and for creating a buffer to store PLC samples in. Even when we
receive a frame with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a basis for when
it comes time to actually do a PLC fill-in.
So, reviewers, now I ask for your help. First off, there's the one line change
in chan_sip that I have put in. Is it right? By my logic it seems correct, but
I'm sure someone can tell me why it is not going to work. This is probably the
change I'm least concerned about, though. What concerns me much more is the
set of changes in channel.c. First off, am I even doing it right? When I run
tests, I can clearly see that when PLC is activated, I see a significant increase
in RTP traffic where I would expect it to be. However, in my humble opinion, the
audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to
me than when no PLC is used at all. I need someone to review the logic I have used
to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic
is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm
sure someone can point out somewhere where I've done something incorrectly.
As I was writing this review request up, I decided to give the code a test run under
valgrind, and I find that for some reason, calls to plc_rx are causing some invalid
reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around
a bit to see why that is the case. If it's obvious to someone reviewing, speak up!
Finally, I have one other proposal that is not reflected in my code review. Since
without transcode_via_sln set, one cannot predict or control where a translation
path will be up, it seems to me that the current practice of using PLC only when
transcoding to SLINEAR is not useful. I recommend that once it has been determined
that the method used in this code review is correct and works as expected, then
the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 21:29:08 +00:00
ast_plc_reload ( ) ;
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 )
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{
unsigned int i ;
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int first = 1 ;
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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 ) ) {
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if ( ! first ) {
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strncat ( buf , " , " , buflen - strlen ( buf ) - 1 ) ;
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} else {
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first = 0 ;
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}
snprintf ( num , sizeof ( num ) , " %u " , i ) ;
2008-03-07 06:54:47 +00:00
strncat ( buf , num , buflen - strlen ( buf ) - 1 ) ;
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}
}
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 ,
2009-11-04 14:05:12 +00:00
. subclass . codec = 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 )
{
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switch ( reason ) {
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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
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
}
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
void ast_connected_line_copy_from_caller ( struct ast_party_connected_line * dest , const struct ast_party_caller * src )
2009-04-03 22:41:46 +00:00
{
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ast_party_id_copy ( & dest - > id , & src - > id ) ;
2010-07-14 16:58:03 +00:00
ast_party_id_copy ( & dest - > ani , & src - > ani ) ;
2009-04-03 22:41:46 +00:00
dest - > ani2 = src - > ani2 ;
}
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void ast_connected_line_copy_to_caller ( struct ast_party_caller * dest , const struct ast_party_connected_line * src )
2009-04-03 22:41:46 +00:00
{
ast_party_id_copy ( & dest - > id , & src - > id ) ;
2010-07-14 16:58:03 +00:00
ast_party_id_copy ( & dest - > ani , & src - > ani ) ;
2009-04-03 22:41:46 +00:00
dest - > ani2 = src - > ani2 ;
}
2010-07-14 15:48:36 +00:00
void ast_channel_set_connected_line ( struct ast_channel * chan , const struct ast_party_connected_line * connected , const struct ast_set_party_connected_line * update )
2009-04-03 22:41:46 +00:00
{
if ( & chan - > connected = = connected ) {
/* Don't set to self */
return ;
}
ast_channel_lock ( chan ) ;
2010-07-14 15:48:36 +00:00
ast_party_connected_line_set ( & chan - > connected , connected , update ) ;
2009-04-03 22:41:46 +00:00
ast_channel_unlock ( chan ) ;
}
2010-07-14 15:48:36 +00:00
/*! \note Should follow struct ast_party_name */
struct ast_party_name_ies {
/*! \brief Subscriber name ie */
int str ;
/*! \brief Character set ie. */
int char_set ;
/*! \brief presentation-indicator ie */
int presentation ;
/*! \brief valid/present ie */
int valid ;
2009-04-03 22:41:46 +00:00
} ;
2010-07-14 15:48:36 +00:00
/*!
* \ internal
* \ since 1.8
* \ brief Build a party name information data frame component .
*
* \ param data Buffer to fill with the frame data
* \ param datalen Size of the buffer to fill
* \ param name Party name information
* \ param label Name of particular party name
* \ param ies Data frame ie values for the party name components
*
* \ retval - 1 if error
* \ retval Amount of data buffer used
*/
static int party_name_build_data ( unsigned char * data , size_t datalen , const struct ast_party_name * name , const char * label , const struct ast_party_name_ies * ies )
2009-04-03 22:41:46 +00:00
{
size_t length ;
size_t pos = 0 ;
/*
* The size of integer values must be fixed in case the frame is
* shipped to another machine .
*/
2010-07-14 15:48:36 +00:00
if ( name - > str ) {
length = strlen ( name - > str ) ;
2009-04-03 22:41:46 +00:00
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + length ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " No space left for %s name \n " , label ) ;
2009-04-03 22:41:46 +00:00
return - 1 ;
}
2010-07-14 15:48:36 +00:00
data [ pos + + ] = ies - > str ;
2009-04-03 22:41:46 +00:00
data [ pos + + ] = length ;
2010-07-14 15:48:36 +00:00
memcpy ( data + pos , name - > str , length ) ;
2009-04-03 22:41:46 +00:00
pos + = length ;
}
2010-07-14 15:48:36 +00:00
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + 1 ) {
ast_log ( LOG_WARNING , " No space left for %s name char set \n " , label ) ;
return - 1 ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
data [ pos + + ] = ies - > char_set ;
data [ pos + + ] = 1 ;
data [ pos + + ] = name - > char_set ;
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + 1 ) {
ast_log ( LOG_WARNING , " No space left for %s name presentation \n " , label ) ;
return - 1 ;
}
data [ pos + + ] = ies - > presentation ;
data [ pos + + ] = 1 ;
data [ pos + + ] = name - > presentation ;
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + 1 ) {
ast_log ( LOG_WARNING , " No space left for %s name valid \n " , label ) ;
return - 1 ;
}
data [ pos + + ] = ies - > valid ;
data [ pos + + ] = 1 ;
data [ pos + + ] = name - > valid ;
return pos ;
}
/*! \note Should follow struct ast_party_number */
struct ast_party_number_ies {
/*! \brief Subscriber phone number ie */
int str ;
/*! \brief Type-Of-Number and Numbering-Plan ie */
int plan ;
/*! \brief presentation-indicator ie */
int presentation ;
/*! \brief valid/present ie */
int valid ;
} ;
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
/*!
* \ internal
* \ since 1.8
* \ brief Build a party number information data frame component .
*
* \ param data Buffer to fill with the frame data
* \ param datalen Size of the buffer to fill
* \ param number Party number information
* \ param label Name of particular party number
* \ param ies Data frame ie values for the party number components
*
* \ retval - 1 if error
* \ retval Amount of data buffer used
*/
static int party_number_build_data ( unsigned char * data , size_t datalen , const struct ast_party_number * number , const char * label , const struct ast_party_number_ies * ies )
{
size_t length ;
size_t pos = 0 ;
/*
* The size of integer values must be fixed in case the frame is
* shipped to another machine .
*/
if ( number - > str ) {
length = strlen ( number - > str ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + length ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " No space left for %s number \n " , label ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
return - 1 ;
}
2010-07-14 15:48:36 +00:00
data [ pos + + ] = ies - > str ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
data [ pos + + ] = length ;
2010-07-14 15:48:36 +00:00
memcpy ( data + pos , number - > str , length ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
pos + = length ;
}
2009-04-03 22:41:46 +00:00
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " No space left for %s numbering plan \n " , label ) ;
2009-04-03 22:41:46 +00:00
return - 1 ;
}
2010-07-14 15:48:36 +00:00
data [ pos + + ] = ies - > plan ;
2009-04-03 22:41:46 +00:00
data [ pos + + ] = 1 ;
2010-07-14 15:48:36 +00:00
data [ pos + + ] = number - > plan ;
2009-04-03 22:41:46 +00:00
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " No space left for %s number presentation \n " , label ) ;
2009-04-03 22:41:46 +00:00
return - 1 ;
}
2010-07-14 15:48:36 +00:00
data [ pos + + ] = ies - > presentation ;
2009-04-03 22:41:46 +00:00
data [ pos + + ] = 1 ;
2010-07-14 15:48:36 +00:00
data [ pos + + ] = number - > presentation ;
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + 1 ) {
ast_log ( LOG_WARNING , " No space left for %s number valid \n " , label ) ;
2009-04-03 22:41:46 +00:00
return - 1 ;
}
2010-07-14 15:48:36 +00:00
data [ pos + + ] = ies - > valid ;
data [ pos + + ] = 1 ;
data [ pos + + ] = number - > valid ;
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
return pos ;
}
/*! \note Should follow struct ast_party_subaddress */
struct ast_party_subaddress_ies {
/*! \brief subaddress ie. */
int str ;
/*! \brief subaddress type ie */
int type ;
/*! \brief odd/even indicator ie */
int odd_even_indicator ;
/*! \brief valid/present ie */
int valid ;
} ;
/*!
* \ internal
* \ since 1.8
* \ brief Build a party subaddress information data frame component .
*
* \ param data Buffer to fill with the frame data
* \ param datalen Size of the buffer to fill
* \ param subaddress Party subaddress information
* \ param label Name of particular party subaddress
* \ param ies Data frame ie values for the party subaddress components
*
* \ retval - 1 if error
* \ retval Amount of data buffer used
*/
static int party_subaddress_build_data ( unsigned char * data , size_t datalen , const struct ast_party_subaddress * subaddress , const char * label , const struct ast_party_subaddress_ies * ies )
{
size_t length ;
size_t pos = 0 ;
/*
* The size of integer values must be fixed in case the frame is
* shipped to another machine .
*/
if ( subaddress - > str ) {
length = strlen ( subaddress - > str ) ;
2009-10-22 16:33:22 +00:00
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + length ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " No space left for %s subaddress \n " , label ) ;
2009-10-22 16:33:22 +00:00
return - 1 ;
}
2010-07-14 15:48:36 +00:00
data [ pos + + ] = ies - > str ;
2009-10-22 16:33:22 +00:00
data [ pos + + ] = length ;
2010-07-14 15:48:36 +00:00
memcpy ( data + pos , subaddress - > str , length ) ;
2009-10-22 16:33:22 +00:00
pos + = length ;
}
2010-07-14 15:48:36 +00:00
2009-10-22 16:33:22 +00:00
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " No space left for %s type of subaddress \n " , label ) ;
2009-10-22 16:33:22 +00:00
return - 1 ;
}
2010-07-14 15:48:36 +00:00
data [ pos + + ] = ies - > type ;
2009-10-22 16:33:22 +00:00
data [ pos + + ] = 1 ;
2010-07-14 15:48:36 +00:00
data [ pos + + ] = subaddress - > type ;
2009-10-22 16:33:22 +00:00
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + 1 ) {
ast_log ( LOG_WARNING ,
2010-07-14 15:48:36 +00:00
" No space left for %s subaddress odd-even indicator \n " , label ) ;
return - 1 ;
}
data [ pos + + ] = ies - > odd_even_indicator ;
data [ pos + + ] = 1 ;
data [ pos + + ] = subaddress - > odd_even_indicator ;
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + 1 ) {
ast_log ( LOG_WARNING , " No space left for %s subaddress valid \n " , label ) ;
2009-10-22 16:33:22 +00:00
return - 1 ;
}
2010-07-14 15:48:36 +00:00
data [ pos + + ] = ies - > valid ;
2009-10-22 16:33:22 +00:00
data [ pos + + ] = 1 ;
2010-07-14 15:48:36 +00:00
data [ pos + + ] = subaddress - > valid ;
return pos ;
}
/*! \note Should follow struct ast_party_id */
struct ast_party_id_ies {
/*! \brief Subscriber name ies */
struct ast_party_name_ies name ;
/*! \brief Subscriber phone number ies */
struct ast_party_number_ies number ;
/*! \brief Subscriber subaddress ies. */
struct ast_party_subaddress_ies subaddress ;
/*! \brief User party id tag ie. */
int tag ;
/*! \brief Combined name and number presentation ie. */
int combined_presentation ;
} ;
/*!
* \ internal
* \ since 1.8
* \ brief Build a party id information data frame component .
*
* \ param data Buffer to fill with the frame data
* \ param datalen Size of the buffer to fill
* \ param id Party id information
* \ param label Name of particular party id
* \ param ies Data frame ie values for the party id components
* \ param update What id information to build . NULL if all .
*
* \ retval - 1 if error
* \ retval Amount of data buffer used
*/
static int party_id_build_data ( unsigned char * data , size_t datalen ,
const struct ast_party_id * id , const char * label , const struct ast_party_id_ies * ies ,
const struct ast_set_party_id * update )
{
size_t length ;
size_t pos = 0 ;
int res ;
2009-10-22 16:33:22 +00:00
2010-07-14 15:48:36 +00:00
/*
* The size of integer values must be fixed in case the frame is
* shipped to another machine .
*/
if ( ! update | | update - > name ) {
res = party_name_build_data ( data + pos , datalen - pos , & id - > name , label ,
& ies - > name ) ;
if ( res < 0 ) {
return - 1 ;
}
pos + = res ;
}
if ( ! update | | update - > number ) {
res = party_number_build_data ( data + pos , datalen - pos , & id - > number , label ,
& ies - > number ) ;
if ( res < 0 ) {
return - 1 ;
}
pos + = res ;
}
if ( ! update | | update - > subaddress ) {
res = party_subaddress_build_data ( data + pos , datalen - pos , & id - > subaddress ,
label , & ies - > subaddress ) ;
if ( res < 0 ) {
return - 1 ;
}
pos + = res ;
}
/* *************** Party id user tag **************************** */
if ( id - > tag ) {
length = strlen ( id - > tag ) ;
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + length ) {
ast_log ( LOG_WARNING , " No space left for %s tag \n " , label ) ;
return - 1 ;
}
data [ pos + + ] = ies - > tag ;
data [ pos + + ] = length ;
memcpy ( data + pos , id - > tag , length ) ;
pos + = length ;
}
/* *************** Party id combined presentation *************** */
if ( ! update | | update - > number ) {
int presentation ;
if ( ! update | | update - > name ) {
presentation = ast_party_id_presentation ( id ) ;
} else {
/*
* We must compromise because not all the information is available
* to determine a combined presentation value .
* We will only send the number presentation instead .
*/
presentation = id - > number . presentation ;
}
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + 1 ) {
ast_log ( LOG_WARNING , " No space left for %s combined presentation \n " , label ) ;
return - 1 ;
}
data [ pos + + ] = ies - > combined_presentation ;
data [ pos + + ] = 1 ;
data [ pos + + ] = presentation ;
}
return pos ;
}
/*!
* \ brief Element identifiers for connected line indication frame data
* \ note Only add to the end of this enum .
*/
enum {
AST_CONNECTED_LINE_NUMBER ,
AST_CONNECTED_LINE_NAME ,
AST_CONNECTED_LINE_NUMBER_PLAN ,
AST_CONNECTED_LINE_ID_PRESENTATION , /* Combined number and name presentation. */
AST_CONNECTED_LINE_SOURCE ,
AST_CONNECTED_LINE_SUBADDRESS ,
AST_CONNECTED_LINE_SUBADDRESS_TYPE ,
AST_CONNECTED_LINE_SUBADDRESS_ODD_EVEN ,
AST_CONNECTED_LINE_SUBADDRESS_VALID ,
AST_CONNECTED_LINE_TAG ,
AST_CONNECTED_LINE_VERSION ,
AST_CONNECTED_LINE_NAME_VALID ,
AST_CONNECTED_LINE_NAME_CHAR_SET ,
AST_CONNECTED_LINE_NAME_PRESENTATION ,
AST_CONNECTED_LINE_NUMBER_VALID ,
AST_CONNECTED_LINE_NUMBER_PRESENTATION ,
} ;
int ast_connected_line_build_data ( unsigned char * data , size_t datalen , const struct ast_party_connected_line * connected , const struct ast_set_party_connected_line * update )
{
int32_t value ;
size_t pos = 0 ;
int res ;
static const struct ast_party_id_ies ies = {
. name . str = AST_CONNECTED_LINE_NAME ,
. name . char_set = AST_CONNECTED_LINE_NAME_CHAR_SET ,
. name . presentation = AST_CONNECTED_LINE_NAME_PRESENTATION ,
. name . valid = AST_CONNECTED_LINE_NAME_VALID ,
. number . str = AST_CONNECTED_LINE_NUMBER ,
. number . plan = AST_CONNECTED_LINE_NUMBER_PLAN ,
. number . presentation = AST_CONNECTED_LINE_NUMBER_PRESENTATION ,
. number . valid = AST_CONNECTED_LINE_NUMBER_VALID ,
. subaddress . str = AST_CONNECTED_LINE_SUBADDRESS ,
. subaddress . type = AST_CONNECTED_LINE_SUBADDRESS_TYPE ,
. subaddress . odd_even_indicator = AST_CONNECTED_LINE_SUBADDRESS_ODD_EVEN ,
. subaddress . valid = AST_CONNECTED_LINE_SUBADDRESS_VALID ,
. tag = AST_CONNECTED_LINE_TAG ,
. combined_presentation = AST_CONNECTED_LINE_ID_PRESENTATION ,
} ;
/*
* The size of integer values must be fixed in case the frame is
* shipped to another machine .
*/
/* Connected line frame version */
2009-10-22 16:33:22 +00:00
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " No space left for connected line frame version \n " ) ;
2009-10-22 16:33:22 +00:00
return - 1 ;
}
2010-07-14 15:48:36 +00:00
data [ pos + + ] = AST_CONNECTED_LINE_VERSION ;
2009-10-22 16:33:22 +00:00
data [ pos + + ] = 1 ;
2010-07-14 15:48:36 +00:00
data [ pos + + ] = 2 ; /* Version 1 did not have a version ie */
res = party_id_build_data ( data + pos , datalen - pos , & connected - > id ,
" connected line " , & ies , update ? & update - > id : NULL ) ;
if ( res < 0 ) {
return - 1 ;
}
pos + = res ;
/* Connected line source */
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + sizeof ( value ) ) {
ast_log ( LOG_WARNING , " No space left for connected line source \n " ) ;
return - 1 ;
}
data [ pos + + ] = AST_CONNECTED_LINE_SOURCE ;
data [ pos + + ] = sizeof ( value ) ;
value = htonl ( connected - > source ) ;
memcpy ( data + pos , & value , sizeof ( value ) ) ;
pos + = sizeof ( value ) ;
2009-10-22 16:33:22 +00:00
2009-04-03 22:41:46 +00:00
return pos ;
}
int ast_connected_line_parse_data ( const unsigned char * data , size_t datalen , struct ast_party_connected_line * connected )
{
size_t pos ;
unsigned char ie_len ;
unsigned char ie_id ;
int32_t value ;
2010-07-14 15:48:36 +00:00
int frame_version = 1 ;
int combined_presentation = 0 ;
int got_combined_presentation = 0 ; /* TRUE if got a combined name and number presentation value. */
2009-04-03 22:41:46 +00:00
for ( pos = 0 ; pos < datalen ; pos + = ie_len ) {
if ( datalen < pos + sizeof ( ie_id ) + sizeof ( ie_len ) ) {
ast_log ( LOG_WARNING , " Invalid connected line update \n " ) ;
return - 1 ;
}
ie_id = data [ pos + + ] ;
ie_len = data [ pos + + ] ;
if ( datalen < pos + ie_len ) {
ast_log ( LOG_WARNING , " Invalid connected line update \n " ) ;
return - 1 ;
}
switch ( ie_id ) {
2010-07-14 15:48:36 +00:00
/* Connected line party frame version */
case AST_CONNECTED_LINE_VERSION :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid connected line frame version (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
frame_version = data [ pos ] ;
2009-04-03 22:41:46 +00:00
break ;
2010-07-14 15:48:36 +00:00
/* Connected line party id name */
2009-04-03 22:41:46 +00:00
case AST_CONNECTED_LINE_NAME :
2010-07-14 15:48:36 +00:00
ast_free ( connected - > id . name . str ) ;
connected - > id . name . str = ast_malloc ( ie_len + 1 ) ;
if ( connected - > id . name . str ) {
memcpy ( connected - > id . name . str , data + pos , ie_len ) ;
connected - > id . name . str [ ie_len ] = 0 ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
break ;
case AST_CONNECTED_LINE_NAME_CHAR_SET :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid connected line name char set (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
connected - > id . name . char_set = data [ pos ] ;
2009-04-03 22:41:46 +00:00
break ;
2010-07-14 15:48:36 +00:00
case AST_CONNECTED_LINE_NAME_PRESENTATION :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid connected line name presentation (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
}
2010-07-14 15:48:36 +00:00
connected - > id . name . presentation = data [ pos ] ;
break ;
case AST_CONNECTED_LINE_NAME_VALID :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid connected line name valid (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
}
connected - > id . name . valid = data [ pos ] ;
break ;
/* Connected line party id number */
case AST_CONNECTED_LINE_NUMBER :
ast_free ( connected - > id . number . str ) ;
connected - > id . number . str = ast_malloc ( ie_len + 1 ) ;
if ( connected - > id . number . str ) {
memcpy ( connected - > id . number . str , data + pos , ie_len ) ;
connected - > id . number . str [ ie_len ] = 0 ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
}
break ;
2010-07-14 15:48:36 +00:00
case AST_CONNECTED_LINE_NUMBER_PLAN :
2009-04-03 22:41:46 +00:00
if ( ie_len ! = 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " Invalid connected line numbering plan (%u) \n " ,
( unsigned ) ie_len ) ;
2009-04-03 22:41:46 +00:00
break ;
}
2010-07-14 15:48:36 +00:00
connected - > id . number . plan = data [ pos ] ;
2009-04-03 22:41:46 +00:00
break ;
case AST_CONNECTED_LINE_NUMBER_PRESENTATION :
if ( ie_len ! = 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " Invalid connected line number presentation (%u) \n " ,
( unsigned ) ie_len ) ;
2009-04-03 22:41:46 +00:00
break ;
}
2010-07-14 15:48:36 +00:00
connected - > id . number . presentation = data [ pos ] ;
2009-04-03 22:41:46 +00:00
break ;
2010-07-14 15:48:36 +00:00
case AST_CONNECTED_LINE_NUMBER_VALID :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid connected line number valid (%u) \n " ,
( unsigned ) ie_len ) ;
2009-04-03 22:41:46 +00:00
break ;
}
2010-07-14 15:48:36 +00:00
connected - > id . number . valid = data [ pos ] ;
2009-04-03 22:41:46 +00:00
break ;
2010-07-14 15:48:36 +00:00
/* Connected line party id combined presentation */
case AST_CONNECTED_LINE_ID_PRESENTATION :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid connected line combined presentation (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
2009-10-22 16:33:22 +00:00
}
2010-07-14 15:48:36 +00:00
combined_presentation = data [ pos ] ;
got_combined_presentation = 1 ;
break ;
/* Connected line party id subaddress */
case AST_CONNECTED_LINE_SUBADDRESS :
ast_free ( connected - > id . subaddress . str ) ;
2009-10-22 16:33:22 +00:00
connected - > id . subaddress . str = ast_malloc ( ie_len + 1 ) ;
if ( connected - > id . subaddress . str ) {
memcpy ( connected - > id . subaddress . str , data + pos , ie_len ) ;
connected - > id . subaddress . str [ ie_len ] = 0 ;
}
break ;
case AST_CONNECTED_LINE_SUBADDRESS_TYPE :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid connected line type of subaddress (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
}
connected - > id . subaddress . type = data [ pos ] ;
break ;
case AST_CONNECTED_LINE_SUBADDRESS_ODD_EVEN :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING ,
" Invalid connected line subaddress odd-even indicator (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
}
connected - > id . subaddress . odd_even_indicator = data [ pos ] ;
break ;
case AST_CONNECTED_LINE_SUBADDRESS_VALID :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid connected line subaddress valid (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
}
connected - > id . subaddress . valid = data [ pos ] ;
break ;
2010-07-14 15:48:36 +00:00
/* Connected line party tag */
case AST_CONNECTED_LINE_TAG :
ast_free ( connected - > id . tag ) ;
connected - > id . tag = ast_malloc ( ie_len + 1 ) ;
if ( connected - > id . tag ) {
memcpy ( connected - > id . tag , data + pos , ie_len ) ;
connected - > id . tag [ ie_len ] = 0 ;
}
break ;
/* Connected line party source */
case AST_CONNECTED_LINE_SOURCE :
if ( ie_len ! = sizeof ( value ) ) {
ast_log ( LOG_WARNING , " Invalid connected line source (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
}
memcpy ( & value , data + pos , sizeof ( value ) ) ;
connected - > source = ntohl ( value ) ;
break ;
/* Connected line party unknown element */
2009-04-03 22:41:46 +00:00
default :
2010-07-14 15:48:36 +00:00
ast_log ( LOG_DEBUG , " Unknown connected line element: %u (%u) \n " ,
( unsigned ) ie_id , ( unsigned ) ie_len ) ;
2009-04-03 22:41:46 +00:00
break ;
}
}
2010-07-14 15:48:36 +00:00
switch ( frame_version ) {
case 1 :
/*
* The other end is an earlier version that we need to adjust
* for compatibility .
*/
connected - > id . name . valid = 1 ;
connected - > id . name . char_set = AST_PARTY_CHAR_SET_ISO8859_1 ;
connected - > id . number . valid = 1 ;
if ( got_combined_presentation ) {
connected - > id . name . presentation = combined_presentation ;
connected - > id . number . presentation = combined_presentation ;
}
break ;
case 2 :
/* The other end is at the same level as we are. */
break ;
default :
/*
* The other end is newer than we are .
* We need to assume that they are compatible with us .
*/
ast_log ( LOG_DEBUG , " Connected line frame has newer version: %u \n " ,
( unsigned ) frame_version ) ;
break ;
}
2009-04-03 22:41:46 +00:00
return 0 ;
}
2010-07-14 15:48:36 +00:00
void ast_channel_update_connected_line ( struct ast_channel * chan , const struct ast_party_connected_line * connected , const struct ast_set_party_connected_line * update )
2009-04-03 22:41:46 +00:00
{
2010-09-21 19:02:40 +00:00
unsigned char data [ 1024 ] ; /* This should be large enough */
2009-04-03 22:41:46 +00:00
size_t datalen ;
2010-07-14 15:48:36 +00:00
datalen = ast_connected_line_build_data ( data , sizeof ( data ) , connected , update ) ;
2009-04-03 22:41:46 +00:00
if ( datalen = = ( size_t ) - 1 ) {
return ;
}
2010-09-21 19:02:40 +00:00
ast_indicate_data ( chan , AST_CONTROL_CONNECTED_LINE , data , datalen ) ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
void ast_channel_queue_connected_line_update ( struct ast_channel * chan , const struct ast_party_connected_line * connected , const struct ast_set_party_connected_line * update )
2009-04-03 22:41:46 +00:00
{
unsigned char data [ 1024 ] ; /* This should be large enough */
size_t datalen ;
2010-07-14 15:48:36 +00:00
datalen = ast_connected_line_build_data ( data , sizeof ( data ) , connected , update ) ;
2009-04-03 22:41:46 +00:00
if ( datalen = = ( size_t ) - 1 ) {
return ;
}
ast_queue_control_data ( chan , AST_CONTROL_CONNECTED_LINE , data , datalen ) ;
}
2010-07-14 15:48:36 +00:00
void ast_channel_set_redirecting ( struct ast_channel * chan , const struct ast_party_redirecting * redirecting , const struct ast_set_party_redirecting * update )
2009-04-03 22:41:46 +00:00
{
if ( & chan - > redirecting = = redirecting ) {
/* Don't set to self */
return ;
}
ast_channel_lock ( chan ) ;
2010-07-14 15:48:36 +00:00
ast_party_redirecting_set ( & chan - > redirecting , redirecting , update ) ;
2009-04-03 22:41:46 +00:00
ast_channel_unlock ( chan ) ;
}
/*!
* \ brief Element identifiers for redirecting indication frame data
* \ note Only add to the end of this enum .
*/
enum {
AST_REDIRECTING_FROM_NUMBER ,
AST_REDIRECTING_FROM_NAME ,
2010-07-14 15:48:36 +00:00
AST_REDIRECTING_FROM_NUMBER_PLAN ,
AST_REDIRECTING_FROM_ID_PRESENTATION ,
2009-04-03 22:41:46 +00:00
AST_REDIRECTING_TO_NUMBER ,
AST_REDIRECTING_TO_NAME ,
2010-07-14 15:48:36 +00:00
AST_REDIRECTING_TO_NUMBER_PLAN ,
AST_REDIRECTING_TO_ID_PRESENTATION ,
2009-04-03 22:41:46 +00:00
AST_REDIRECTING_REASON ,
2009-10-22 16:33:22 +00:00
AST_REDIRECTING_COUNT ,
AST_REDIRECTING_FROM_SUBADDRESS ,
AST_REDIRECTING_FROM_SUBADDRESS_TYPE ,
AST_REDIRECTING_FROM_SUBADDRESS_ODD_EVEN ,
AST_REDIRECTING_FROM_SUBADDRESS_VALID ,
AST_REDIRECTING_TO_SUBADDRESS ,
AST_REDIRECTING_TO_SUBADDRESS_TYPE ,
AST_REDIRECTING_TO_SUBADDRESS_ODD_EVEN ,
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
AST_REDIRECTING_TO_SUBADDRESS_VALID ,
AST_REDIRECTING_FROM_TAG ,
AST_REDIRECTING_TO_TAG ,
2010-07-14 15:48:36 +00:00
AST_REDIRECTING_VERSION ,
AST_REDIRECTING_FROM_NAME_VALID ,
AST_REDIRECTING_FROM_NAME_CHAR_SET ,
AST_REDIRECTING_FROM_NAME_PRESENTATION ,
AST_REDIRECTING_FROM_NUMBER_VALID ,
AST_REDIRECTING_FROM_NUMBER_PRESENTATION ,
AST_REDIRECTING_TO_NAME_VALID ,
AST_REDIRECTING_TO_NAME_CHAR_SET ,
AST_REDIRECTING_TO_NAME_PRESENTATION ,
AST_REDIRECTING_TO_NUMBER_VALID ,
AST_REDIRECTING_TO_NUMBER_PRESENTATION ,
2009-04-03 22:41:46 +00:00
} ;
2010-07-14 15:48:36 +00:00
int ast_redirecting_build_data ( unsigned char * data , size_t datalen , const struct ast_party_redirecting * redirecting , const struct ast_set_party_redirecting * update )
2009-04-03 22:41:46 +00:00
{
int32_t value ;
size_t pos = 0 ;
2010-07-14 15:48:36 +00:00
int res ;
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
static const struct ast_party_id_ies from_ies = {
. name . str = AST_REDIRECTING_FROM_NAME ,
. name . char_set = AST_REDIRECTING_FROM_NAME_CHAR_SET ,
. name . presentation = AST_REDIRECTING_FROM_NAME_PRESENTATION ,
. name . valid = AST_REDIRECTING_FROM_NAME_VALID ,
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
2010-07-14 15:48:36 +00:00
. number . str = AST_REDIRECTING_FROM_NUMBER ,
. number . plan = AST_REDIRECTING_FROM_NUMBER_PLAN ,
. number . presentation = AST_REDIRECTING_FROM_NUMBER_PRESENTATION ,
. number . valid = AST_REDIRECTING_FROM_NUMBER_VALID ,
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
. subaddress . str = AST_REDIRECTING_FROM_SUBADDRESS ,
. subaddress . type = AST_REDIRECTING_FROM_SUBADDRESS_TYPE ,
. subaddress . odd_even_indicator = AST_REDIRECTING_FROM_SUBADDRESS_ODD_EVEN ,
. subaddress . valid = AST_REDIRECTING_FROM_SUBADDRESS_VALID ,
2009-04-03 22:41:46 +00:00
2010-07-14 15:48:36 +00:00
. tag = AST_REDIRECTING_FROM_TAG ,
. combined_presentation = AST_REDIRECTING_FROM_ID_PRESENTATION ,
} ;
static const struct ast_party_id_ies to_ies = {
. name . str = AST_REDIRECTING_TO_NAME ,
. name . char_set = AST_REDIRECTING_TO_NAME_CHAR_SET ,
. name . presentation = AST_REDIRECTING_TO_NAME_PRESENTATION ,
. name . valid = AST_REDIRECTING_TO_NAME_VALID ,
. number . str = AST_REDIRECTING_TO_NUMBER ,
. number . plan = AST_REDIRECTING_TO_NUMBER_PLAN ,
. number . presentation = AST_REDIRECTING_TO_NUMBER_PRESENTATION ,
. number . valid = AST_REDIRECTING_TO_NUMBER_VALID ,
. subaddress . str = AST_REDIRECTING_TO_SUBADDRESS ,
. subaddress . type = AST_REDIRECTING_TO_SUBADDRESS_TYPE ,
. subaddress . odd_even_indicator = AST_REDIRECTING_TO_SUBADDRESS_ODD_EVEN ,
. subaddress . valid = AST_REDIRECTING_TO_SUBADDRESS_VALID ,
. tag = AST_REDIRECTING_TO_TAG ,
. combined_presentation = AST_REDIRECTING_TO_ID_PRESENTATION ,
} ;
2009-10-22 16:33:22 +00:00
2010-07-14 15:48:36 +00:00
/* Redirecting frame version */
2009-10-22 16:33:22 +00:00
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " No space left for redirecting frame version \n " ) ;
2009-10-22 16:33:22 +00:00
return - 1 ;
}
2010-07-14 15:48:36 +00:00
data [ pos + + ] = AST_REDIRECTING_VERSION ;
2009-10-22 16:33:22 +00:00
data [ pos + + ] = 1 ;
2010-07-14 15:48:36 +00:00
data [ pos + + ] = 2 ; /* Version 1 did not have a version ie */
2009-10-22 16:33:22 +00:00
2010-07-14 15:48:36 +00:00
res = party_id_build_data ( data + pos , datalen - pos , & redirecting - > from ,
" redirecting-from " , & from_ies , update ? & update - > from : NULL ) ;
if ( res < 0 ) {
2009-10-22 16:33:22 +00:00
return - 1 ;
}
2010-07-14 15:48:36 +00:00
pos + = res ;
2009-10-22 16:33:22 +00:00
2010-07-14 15:48:36 +00:00
res = party_id_build_data ( data + pos , datalen - pos , & redirecting - > to ,
" redirecting-to " , & to_ies , update ? & update - > to : NULL ) ;
if ( res < 0 ) {
2009-10-22 16:33:22 +00:00
return - 1 ;
}
2010-07-14 15:48:36 +00:00
pos + = res ;
2009-10-22 16:33:22 +00:00
2009-04-03 22:41:46 +00:00
/* Redirecting reason */
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + sizeof ( value ) ) {
ast_log ( LOG_WARNING , " No space left for redirecting reason \n " ) ;
return - 1 ;
}
data [ pos + + ] = AST_REDIRECTING_REASON ;
data [ pos + + ] = sizeof ( value ) ;
value = htonl ( redirecting - > reason ) ;
memcpy ( data + pos , & value , sizeof ( value ) ) ;
pos + = sizeof ( value ) ;
/* Redirecting count */
if ( datalen < pos + ( sizeof ( data [ 0 ] ) * 2 ) + sizeof ( value ) ) {
ast_log ( LOG_WARNING , " No space left for redirecting count \n " ) ;
return - 1 ;
}
data [ pos + + ] = AST_REDIRECTING_COUNT ;
data [ pos + + ] = sizeof ( value ) ;
value = htonl ( redirecting - > count ) ;
memcpy ( data + pos , & value , sizeof ( value ) ) ;
pos + = sizeof ( value ) ;
return pos ;
}
int ast_redirecting_parse_data ( const unsigned char * data , size_t datalen , struct ast_party_redirecting * redirecting )
{
size_t pos ;
unsigned char ie_len ;
unsigned char ie_id ;
int32_t value ;
2010-07-14 15:48:36 +00:00
int frame_version = 1 ;
int from_combined_presentation = 0 ;
int got_from_combined_presentation = 0 ; /* TRUE if got a combined name and number presentation value. */
int to_combined_presentation = 0 ;
int got_to_combined_presentation = 0 ; /* TRUE if got a combined name and number presentation value. */
2009-04-03 22:41:46 +00:00
for ( pos = 0 ; pos < datalen ; pos + = ie_len ) {
if ( datalen < pos + sizeof ( ie_id ) + sizeof ( ie_len ) ) {
ast_log ( LOG_WARNING , " Invalid redirecting update \n " ) ;
return - 1 ;
}
ie_id = data [ pos + + ] ;
ie_len = data [ pos + + ] ;
if ( datalen < pos + ie_len ) {
ast_log ( LOG_WARNING , " Invalid redirecting update \n " ) ;
return - 1 ;
}
switch ( ie_id ) {
2010-07-14 15:48:36 +00:00
/* Redirecting frame version */
case AST_REDIRECTING_VERSION :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid redirecting frame version (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
frame_version = data [ pos ] ;
2009-04-03 22:41:46 +00:00
break ;
2010-07-14 15:48:36 +00:00
/* Redirecting-from party id name */
2009-04-03 22:41:46 +00:00
case AST_REDIRECTING_FROM_NAME :
2010-07-14 15:48:36 +00:00
ast_free ( redirecting - > from . name . str ) ;
redirecting - > from . name . str = ast_malloc ( ie_len + 1 ) ;
if ( redirecting - > from . name . str ) {
memcpy ( redirecting - > from . name . str , data + pos , ie_len ) ;
redirecting - > from . name . str [ ie_len ] = 0 ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
break ;
case AST_REDIRECTING_FROM_NAME_CHAR_SET :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid redirecting-from name char set (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
redirecting - > from . name . char_set = data [ pos ] ;
2009-04-03 22:41:46 +00:00
break ;
2010-07-14 15:48:36 +00:00
case AST_REDIRECTING_FROM_NAME_PRESENTATION :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid redirecting-from name presentation (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
}
2010-07-14 15:48:36 +00:00
redirecting - > from . name . presentation = data [ pos ] ;
break ;
case AST_REDIRECTING_FROM_NAME_VALID :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid redirecting-from name valid (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
}
redirecting - > from . name . valid = data [ pos ] ;
break ;
/* Redirecting-from party id number */
case AST_REDIRECTING_FROM_NUMBER :
ast_free ( redirecting - > from . number . str ) ;
redirecting - > from . number . str = ast_malloc ( ie_len + 1 ) ;
if ( redirecting - > from . number . str ) {
memcpy ( redirecting - > from . number . str , data + pos , ie_len ) ;
redirecting - > from . number . str [ ie_len ] = 0 ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
}
break ;
2010-07-14 15:48:36 +00:00
case AST_REDIRECTING_FROM_NUMBER_PLAN :
2009-04-03 22:41:46 +00:00
if ( ie_len ! = 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " Invalid redirecting-from numbering plan (%u) \n " ,
( unsigned ) ie_len ) ;
2009-04-03 22:41:46 +00:00
break ;
}
2010-07-14 15:48:36 +00:00
redirecting - > from . number . plan = data [ pos ] ;
2009-04-03 22:41:46 +00:00
break ;
case AST_REDIRECTING_FROM_NUMBER_PRESENTATION :
if ( ie_len ! = 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " Invalid redirecting-from number presentation (%u) \n " ,
( unsigned ) ie_len ) ;
2009-04-03 22:41:46 +00:00
break ;
}
2010-07-14 15:48:36 +00:00
redirecting - > from . number . presentation = data [ pos ] ;
2009-04-03 22:41:46 +00:00
break ;
2010-07-14 15:48:36 +00:00
case AST_REDIRECTING_FROM_NUMBER_VALID :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid redirecting-from number valid (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
2009-10-22 16:33:22 +00:00
}
2010-07-14 15:48:36 +00:00
redirecting - > from . number . valid = data [ pos ] ;
break ;
/* Redirecting-from party id combined presentation */
case AST_REDIRECTING_FROM_ID_PRESENTATION :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid redirecting-from combined presentation (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
}
from_combined_presentation = data [ pos ] ;
got_from_combined_presentation = 1 ;
break ;
/* Redirecting-from party id subaddress */
case AST_REDIRECTING_FROM_SUBADDRESS :
ast_free ( redirecting - > from . subaddress . str ) ;
2009-10-22 16:33:22 +00:00
redirecting - > from . subaddress . str = ast_malloc ( ie_len + 1 ) ;
if ( redirecting - > from . subaddress . str ) {
memcpy ( redirecting - > from . subaddress . str , data + pos , ie_len ) ;
redirecting - > from . subaddress . str [ ie_len ] = 0 ;
}
break ;
case AST_REDIRECTING_FROM_SUBADDRESS_TYPE :
if ( ie_len ! = 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " Invalid redirecting-from type of subaddress (%u) \n " ,
2009-10-22 16:33:22 +00:00
( unsigned ) ie_len ) ;
break ;
}
redirecting - > from . subaddress . type = data [ pos ] ;
break ;
case AST_REDIRECTING_FROM_SUBADDRESS_ODD_EVEN :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING ,
2010-07-14 15:48:36 +00:00
" Invalid redirecting-from subaddress odd-even indicator (%u) \n " ,
2009-10-22 16:33:22 +00:00
( unsigned ) ie_len ) ;
break ;
}
redirecting - > from . subaddress . odd_even_indicator = data [ pos ] ;
break ;
case AST_REDIRECTING_FROM_SUBADDRESS_VALID :
if ( ie_len ! = 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " Invalid redirecting-from subaddress valid (%u) \n " ,
2009-10-22 16:33:22 +00:00
( unsigned ) ie_len ) ;
break ;
}
redirecting - > from . subaddress . valid = data [ pos ] ;
break ;
2010-07-14 15:48:36 +00:00
/* Redirecting-from party id tag */
case AST_REDIRECTING_FROM_TAG :
ast_free ( redirecting - > from . tag ) ;
redirecting - > from . tag = ast_malloc ( ie_len + 1 ) ;
if ( redirecting - > from . tag ) {
memcpy ( redirecting - > from . tag , data + pos , ie_len ) ;
redirecting - > from . tag [ ie_len ] = 0 ;
2009-04-03 22:41:46 +00:00
}
break ;
2010-07-14 15:48:36 +00:00
/* Redirecting-to party id name */
2009-04-03 22:41:46 +00:00
case AST_REDIRECTING_TO_NAME :
2010-07-14 15:48:36 +00:00
ast_free ( redirecting - > to . name . str ) ;
redirecting - > to . name . str = ast_malloc ( ie_len + 1 ) ;
if ( redirecting - > to . name . str ) {
memcpy ( redirecting - > to . name . str , data + pos , ie_len ) ;
redirecting - > to . name . str [ ie_len ] = 0 ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
break ;
case AST_REDIRECTING_TO_NAME_CHAR_SET :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid redirecting-to name char set (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
2009-04-03 22:41:46 +00:00
}
2010-07-14 15:48:36 +00:00
redirecting - > to . name . char_set = data [ pos ] ;
2009-04-03 22:41:46 +00:00
break ;
2010-07-14 15:48:36 +00:00
case AST_REDIRECTING_TO_NAME_PRESENTATION :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid redirecting-to name presentation (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
}
2010-07-14 15:48:36 +00:00
redirecting - > to . name . presentation = data [ pos ] ;
break ;
case AST_REDIRECTING_TO_NAME_VALID :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid redirecting-to name valid (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
}
2010-07-14 15:48:36 +00:00
redirecting - > to . name . valid = data [ pos ] ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
break ;
2010-07-14 15:48:36 +00:00
/* Redirecting-to party id number */
case AST_REDIRECTING_TO_NUMBER :
ast_free ( redirecting - > to . number . str ) ;
redirecting - > to . number . str = ast_malloc ( ie_len + 1 ) ;
if ( redirecting - > to . number . str ) {
memcpy ( redirecting - > to . number . str , data + pos , ie_len ) ;
redirecting - > to . number . str [ ie_len ] = 0 ;
}
break ;
case AST_REDIRECTING_TO_NUMBER_PLAN :
2009-04-03 22:41:46 +00:00
if ( ie_len ! = 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " Invalid redirecting-to numbering plan (%u) \n " ,
( unsigned ) ie_len ) ;
2009-04-03 22:41:46 +00:00
break ;
}
2010-07-14 15:48:36 +00:00
redirecting - > to . number . plan = data [ pos ] ;
2009-04-03 22:41:46 +00:00
break ;
case AST_REDIRECTING_TO_NUMBER_PRESENTATION :
if ( ie_len ! = 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " Invalid redirecting-to number presentation (%u) \n " ,
( unsigned ) ie_len ) ;
2009-04-03 22:41:46 +00:00
break ;
}
2010-07-14 15:48:36 +00:00
redirecting - > to . number . presentation = data [ pos ] ;
2009-04-03 22:41:46 +00:00
break ;
2010-07-14 15:48:36 +00:00
case AST_REDIRECTING_TO_NUMBER_VALID :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid redirecting-to number valid (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
}
redirecting - > to . number . valid = data [ pos ] ;
break ;
/* Redirecting-to party id combined presentation */
case AST_REDIRECTING_TO_ID_PRESENTATION :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING , " Invalid redirecting-to combined presentation (%u) \n " ,
( unsigned ) ie_len ) ;
break ;
2009-10-22 16:33:22 +00:00
}
2010-07-14 15:48:36 +00:00
to_combined_presentation = data [ pos ] ;
got_to_combined_presentation = 1 ;
break ;
/* Redirecting-to party id subaddress */
case AST_REDIRECTING_TO_SUBADDRESS :
ast_free ( redirecting - > to . subaddress . str ) ;
2009-10-22 16:33:22 +00:00
redirecting - > to . subaddress . str = ast_malloc ( ie_len + 1 ) ;
if ( redirecting - > to . subaddress . str ) {
memcpy ( redirecting - > to . subaddress . str , data + pos , ie_len ) ;
redirecting - > to . subaddress . str [ ie_len ] = 0 ;
}
break ;
case AST_REDIRECTING_TO_SUBADDRESS_TYPE :
if ( ie_len ! = 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " Invalid redirecting-to type of subaddress (%u) \n " ,
2009-10-22 16:33:22 +00:00
( unsigned ) ie_len ) ;
break ;
}
redirecting - > to . subaddress . type = data [ pos ] ;
break ;
case AST_REDIRECTING_TO_SUBADDRESS_ODD_EVEN :
if ( ie_len ! = 1 ) {
ast_log ( LOG_WARNING ,
2010-07-14 15:48:36 +00:00
" Invalid redirecting-to subaddress odd-even indicator (%u) \n " ,
2009-10-22 16:33:22 +00:00
( unsigned ) ie_len ) ;
break ;
}
redirecting - > to . subaddress . odd_even_indicator = data [ pos ] ;
break ;
case AST_REDIRECTING_TO_SUBADDRESS_VALID :
if ( ie_len ! = 1 ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " Invalid redirecting-to subaddress valid (%u) \n " ,
2009-10-22 16:33:22 +00:00
( unsigned ) ie_len ) ;
break ;
}
redirecting - > to . subaddress . valid = data [ pos ] ;
break ;
2010-07-14 15:48:36 +00:00
/* Redirecting-to party id tag */
case AST_REDIRECTING_TO_TAG :
ast_free ( redirecting - > to . tag ) ;
redirecting - > to . tag = ast_malloc ( ie_len + 1 ) ;
if ( redirecting - > to . tag ) {
memcpy ( redirecting - > to . tag , data + pos , ie_len ) ;
redirecting - > to . tag [ ie_len ] = 0 ;
}
break ;
/* Redirecting reason */
2009-04-03 22:41:46 +00:00
case AST_REDIRECTING_REASON :
if ( ie_len ! = sizeof ( value ) ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " Invalid redirecting reason (%u) \n " ,
( unsigned ) ie_len ) ;
2009-04-03 22:41:46 +00:00
break ;
}
memcpy ( & value , data + pos , sizeof ( value ) ) ;
redirecting - > reason = ntohl ( value ) ;
break ;
2010-07-14 15:48:36 +00:00
/* Redirecting count */
2009-04-03 22:41:46 +00:00
case AST_REDIRECTING_COUNT :
if ( ie_len ! = sizeof ( value ) ) {
2010-07-14 15:48:36 +00:00
ast_log ( LOG_WARNING , " Invalid redirecting count (%u) \n " ,
( unsigned ) ie_len ) ;
2009-04-03 22:41:46 +00:00
break ;
}
memcpy ( & value , data + pos , sizeof ( value ) ) ;
redirecting - > count = ntohl ( value ) ;
break ;
2010-07-14 15:48:36 +00:00
/* Redirecting unknown element */
2009-04-03 22:41:46 +00:00
default :
2010-07-14 15:48:36 +00:00
ast_log ( LOG_DEBUG , " Unknown redirecting element: %u (%u) \n " ,
( unsigned ) ie_id , ( unsigned ) ie_len ) ;
2009-04-03 22:41:46 +00:00
break ;
}
}
2010-07-14 15:48:36 +00:00
switch ( frame_version ) {
case 1 :
/*
* The other end is an earlier version that we need to adjust
* for compatibility .
*/
redirecting - > from . name . valid = 1 ;
redirecting - > from . name . char_set = AST_PARTY_CHAR_SET_ISO8859_1 ;
redirecting - > from . number . valid = 1 ;
if ( got_from_combined_presentation ) {
redirecting - > from . name . presentation = from_combined_presentation ;
redirecting - > from . number . presentation = from_combined_presentation ;
}
redirecting - > to . name . valid = 1 ;
redirecting - > to . name . char_set = AST_PARTY_CHAR_SET_ISO8859_1 ;
redirecting - > to . number . valid = 1 ;
if ( got_to_combined_presentation ) {
redirecting - > to . name . presentation = to_combined_presentation ;
redirecting - > to . number . presentation = to_combined_presentation ;
}
break ;
case 2 :
/* The other end is at the same level as we are. */
break ;
default :
/*
* The other end is newer than we are .
* We need to assume that they are compatible with us .
*/
ast_log ( LOG_DEBUG , " Redirecting frame has newer version: %u \n " ,
( unsigned ) frame_version ) ;
break ;
}
2009-04-03 22:41:46 +00:00
return 0 ;
}
2010-07-14 15:48:36 +00:00
void ast_channel_update_redirecting ( struct ast_channel * chan , const struct ast_party_redirecting * redirecting , const struct ast_set_party_redirecting * update )
2009-04-03 22:41:46 +00:00
{
unsigned char data [ 1024 ] ; /* This should be large enough */
size_t datalen ;
2010-07-14 15:48:36 +00:00
datalen = ast_redirecting_build_data ( data , sizeof ( data ) , redirecting , update ) ;
2009-04-03 22:41:46 +00:00
if ( datalen = = ( size_t ) - 1 ) {
return ;
}
ast_indicate_data ( chan , AST_CONTROL_REDIRECTING , data , datalen ) ;
}
2010-07-14 15:48:36 +00:00
void ast_channel_queue_redirecting_update ( struct ast_channel * chan , const struct ast_party_redirecting * redirecting , const struct ast_set_party_redirecting * update )
2009-04-03 22:41:46 +00:00
{
unsigned char data [ 1024 ] ; /* This should be large enough */
size_t datalen ;
2010-07-14 15:48:36 +00:00
datalen = ast_redirecting_build_data ( data , sizeof ( data ) , redirecting , update ) ;
2009-04-03 22:41:46 +00:00
if ( datalen = = ( size_t ) - 1 ) {
return ;
}
ast_queue_control_data ( chan , AST_CONTROL_REDIRECTING , data , datalen ) ;
}
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
int ast_channel_connected_line_macro ( struct ast_channel * autoservice_chan , struct ast_channel * macro_chan , const void * connected_info , int is_caller , int is_frame )
2009-06-01 20:57:31 +00:00
{
const char * macro ;
const char * macro_args ;
int retval ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
ast_channel_lock ( macro_chan ) ;
macro = pbx_builtin_getvar_helper ( macro_chan , is_caller
2010-05-25 16:23:51 +00:00
? " CONNECTED_LINE_CALLER_SEND_MACRO " : " CONNECTED_LINE_CALLEE_SEND_MACRO " ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
macro = ast_strdupa ( S_OR ( macro , " " ) ) ;
macro_args = pbx_builtin_getvar_helper ( macro_chan , is_caller
2010-05-26 20:04:51 +00:00
? " CONNECTED_LINE_CALLER_SEND_MACRO_ARGS " : " CONNECTED_LINE_CALLEE_SEND_MACRO_ARGS " ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
macro_args = ast_strdupa ( S_OR ( macro_args , " " ) ) ;
if ( ast_strlen_zero ( macro ) ) {
2010-09-21 20:29:51 +00:00
ast_channel_unlock ( macro_chan ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
return - 1 ;
}
if ( is_frame ) {
const struct ast_frame * frame = connected_info ;
2010-05-25 16:23:51 +00:00
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
ast_connected_line_parse_data ( frame - > data . ptr , frame - > datalen , & macro_chan - > connected ) ;
2009-06-01 20:57:31 +00:00
} else {
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
const struct ast_party_connected_line * connected = connected_info ;
2010-05-25 16:23:51 +00:00
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
ast_party_connected_line_copy ( & macro_chan - > connected , connected ) ;
2009-06-01 20:57:31 +00:00
}
2010-09-21 20:29:51 +00:00
ast_channel_unlock ( macro_chan ) ;
2009-06-01 20:57:31 +00:00
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
if ( ! ( retval = ast_app_run_macro ( autoservice_chan , macro_chan , macro , macro_args ) ) ) {
2010-09-21 20:29:51 +00:00
ast_channel_lock ( macro_chan ) ;
2010-07-14 15:48:36 +00:00
ast_channel_update_connected_line ( macro_chan , & macro_chan - > connected , NULL ) ;
2010-09-21 20:29:51 +00:00
ast_channel_unlock ( macro_chan ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
}
return retval ;
}
int ast_channel_redirecting_macro ( struct ast_channel * autoservice_chan , struct ast_channel * macro_chan , const void * redirecting_info , int is_caller , int is_frame )
{
const char * macro ;
const char * macro_args ;
int retval ;
2009-06-01 20:57:31 +00:00
ast_channel_lock ( macro_chan ) ;
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
macro = pbx_builtin_getvar_helper ( macro_chan , is_caller
? " REDIRECTING_CALLER_SEND_MACRO " : " REDIRECTING_CALLEE_SEND_MACRO " ) ;
macro = ast_strdupa ( S_OR ( macro , " " ) ) ;
macro_args = pbx_builtin_getvar_helper ( macro_chan , is_caller
? " REDIRECTING_CALLER_SEND_MACRO_ARGS " : " REDIRECTING_CALLEE_SEND_MACRO_ARGS " ) ;
macro_args = ast_strdupa ( S_OR ( macro_args , " " ) ) ;
2009-06-01 20:57:31 +00:00
if ( ast_strlen_zero ( macro ) ) {
2010-09-21 20:29:51 +00:00
ast_channel_unlock ( macro_chan ) ;
2009-06-01 20:57:31 +00:00
return - 1 ;
}
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
if ( is_frame ) {
const struct ast_frame * frame = redirecting_info ;
ast_redirecting_parse_data ( frame - > data . ptr , frame - > datalen , & macro_chan - > redirecting ) ;
2009-06-01 20:57:31 +00:00
} else {
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
const struct ast_party_redirecting * redirecting = redirecting_info ;
ast_party_redirecting_copy ( & macro_chan - > redirecting , redirecting ) ;
2009-06-01 20:57:31 +00:00
}
2010-09-21 20:29:51 +00:00
ast_channel_unlock ( macro_chan ) ;
2009-06-01 20:57:31 +00:00
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
retval = ast_app_run_macro ( autoservice_chan , macro_chan , macro , macro_args ) ;
if ( ! retval ) {
2010-09-21 20:29:51 +00:00
ast_channel_lock ( macro_chan ) ;
2010-07-14 15:48:36 +00:00
ast_channel_update_redirecting ( macro_chan , & macro_chan - > redirecting , NULL ) ;
2010-09-21 20:29:51 +00:00
ast_channel_unlock ( macro_chan ) ;
2009-06-01 20:57:31 +00:00
}
return retval ;
}
Merge Call completion support into trunk.
From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:
1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
multiple calls to the same device. This proved to not be such a good idea
when implementing protocol-specific monitors, and so we ended up using one
monitor per-device per-call.
3. There are some configuration options which were conceived after the document
was written. These are documented in the ccss.conf.sample that is on this
review request.
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.
This implements CCBS and CCNR in several flavors.
First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.
Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:
* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
what is defined in the referenced draft.
* Implementation of the draft required support for SIP PUBLISH. I attempted to write
this in a generic-enough fashion such that if someone were to want to write PUBLISH
support for other event packages, such as dialog-state or presence, most of the effort
would be in writing callbacks specific to the event package.
* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
parser. The PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly. The rest of the
PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
code. In other words, while there is PIDF support here, it is not in any state
where it could easily be applied to other event packages as is.
Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.
Review: https://reviewboard.asterisk.org/r/523
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09 15:31:32 +00:00
static void * channel_cc_params_copy ( void * data )
{
const struct ast_cc_config_params * src = data ;
struct ast_cc_config_params * dest = ast_cc_config_params_init ( ) ;
if ( ! dest ) {
return NULL ;
}
ast_cc_copy_config_params ( dest , src ) ;
return dest ;
}
static void channel_cc_params_destroy ( void * data )
{
struct ast_cc_config_params * cc_params = data ;
ast_cc_config_params_destroy ( cc_params ) ;
}
static const struct ast_datastore_info cc_channel_datastore_info = {
. type = " Call Completion " ,
. duplicate = channel_cc_params_copy ,
. destroy = channel_cc_params_destroy ,
} ;
int ast_channel_cc_params_init ( struct ast_channel * chan ,
const struct ast_cc_config_params * base_params )
{
struct ast_cc_config_params * cc_params ;
struct ast_datastore * cc_datastore ;
if ( ! ( cc_params = ast_cc_config_params_init ( ) ) ) {
return - 1 ;
}
if ( ! ( cc_datastore = ast_datastore_alloc ( & cc_channel_datastore_info , NULL ) ) ) {
ast_cc_config_params_destroy ( cc_params ) ;
return - 1 ;
}
if ( base_params ) {
ast_cc_copy_config_params ( cc_params , base_params ) ;
}
cc_datastore - > data = cc_params ;
ast_channel_datastore_add ( chan , cc_datastore ) ;
return 0 ;
}
struct ast_cc_config_params * ast_channel_get_cc_config_params ( struct ast_channel * chan )
{
struct ast_datastore * cc_datastore ;
if ( ! ( cc_datastore = ast_channel_datastore_find ( chan , & cc_channel_datastore_info , NULL ) ) ) {
/* If we can't find the datastore, it almost definitely means that the channel type being
* used has not had its driver modified to parse CC config parameters . The best action
* to take here is to create the parameters on the spot with the defaults set .
*/
if ( ast_channel_cc_params_init ( chan , NULL ) ) {
return NULL ;
}
if ( ! ( cc_datastore = ast_channel_datastore_find ( chan , & cc_channel_datastore_info , NULL ) ) ) {
/* Should be impossible */
return NULL ;
}
}
ast_assert ( cc_datastore - > data ! = NULL ) ;
return cc_datastore - > data ;
}
int ast_channel_get_device_name ( struct ast_channel * chan , char * device_name , size_t name_buffer_length )
{
int len = name_buffer_length ;
char * dash ;
if ( ! ast_channel_queryoption ( chan , AST_OPTION_DEVICE_NAME , device_name , & len , 0 ) ) {
return 0 ;
}
/* Dang. Do it the old-fashioned way */
ast_copy_string ( device_name , chan - > name , name_buffer_length ) ;
if ( ( dash = strrchr ( device_name , ' - ' ) ) ) {
* dash = ' \0 ' ;
}
return 0 ;
}
int ast_channel_get_cc_agent_type ( struct ast_channel * chan , char * agent_type , size_t size )
{
int len = size ;
char * slash ;
if ( ! ast_channel_queryoption ( chan , AST_OPTION_CC_AGENT_TYPE , agent_type , & len , 0 ) ) {
return 0 ;
}
ast_copy_string ( agent_type , chan - > name , size ) ;
if ( ( slash = strchr ( agent_type , ' / ' ) ) ) {
* slash = ' \0 ' ;
}
return 0 ;
}
2009-05-05 10:34:19 +00:00
/* DO NOT PUT ADDITIONAL FUNCTIONS BELOW THIS BOUNDARY
*
* ONLY FUNCTIONS FOR PROVIDING BACKWARDS ABI COMPATIBILITY BELONG HERE
*
*/
/* Provide binary compatibility for modules that call ast_channel_alloc() directly;
* newly compiled modules will call __ast_channel_alloc ( ) via the macros in channel . h
*/
# undef ast_channel_alloc
2009-06-26 15:28:53 +00:00
struct ast_channel __attribute__ ( ( format ( printf , 10 , 11 ) ) )
2009-05-05 10:34:19 +00:00
* ast_channel_alloc ( int needqueue , int state , const char * cid_num ,
const char * cid_name , const char * acctcode ,
const char * exten , const char * context ,
2009-06-26 15:28:53 +00:00
const char * linkedid , const int amaflag ,
const char * name_fmt , . . . ) ;
2009-05-05 10:34:19 +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 ,
2009-06-26 15:28:53 +00:00
const char * linkedid , const int amaflag ,
const char * name_fmt , . . . )
2009-05-05 10:34:19 +00:00
{
va_list ap1 , ap2 ;
struct ast_channel * result ;
va_start ( ap1 , name_fmt ) ;
va_start ( ap2 , name_fmt ) ;
result = __ast_channel_alloc_ap ( needqueue , state , cid_num , cid_name , acctcode , exten , context ,
2009-06-26 15:28:53 +00:00
linkedid , amaflag , __FILE__ , __LINE__ , __FUNCTION__ , name_fmt , ap1 , ap2 ) ;
2009-05-05 10:34:19 +00:00
va_end ( ap1 ) ;
va_end ( ap2 ) ;
return result ;
}
