1999-12-04 21:35:07 +00:00
/*
2005-09-14 20:46:50 +00:00
* Asterisk - - An open source telephony toolkit .
1999-12-04 21:35:07 +00:00
*
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
* Copyright ( C ) 1999 - 2008 , Digium , Inc .
1999-12-04 21:35:07 +00:00
*
2004-09-22 05:19:06 +00:00
* Mark Spencer < markster @ digium . com >
1999-12-04 21:35:07 +00:00
*
2005-09-14 20:46:50 +00:00
* 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 .
*
1999-12-04 21:35:07 +00:00
* This program is free software , distributed under the terms of
2005-09-14 20:46:50 +00:00
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree .
*/
2005-10-24 20:12:06 +00:00
/*! \file
2005-09-14 20:46:50 +00:00
*
2005-10-24 20:12:06 +00:00
* \ brief dial ( ) & retrydial ( ) - Trivial application to dial a channel and send an URL on answer
2005-12-30 21:18:06 +00:00
*
* \ author Mark Spencer < markster @ digium . com >
2008-02-09 11:27:10 +00:00
*
2005-11-06 15:09:47 +00:00
* \ ingroup applications
1999-12-04 21:35:07 +00:00
*/
2008-01-22 17:42:27 +00:00
/*** MODULEINFO
2008-04-25 20:20:10 +00:00
< depend > chan_local < / depend >
2008-01-22 17:42:27 +00:00
* * */
2006-06-07 18:54:56 +00:00
# include "asterisk.h"
ASTERISK_FILE_VERSION ( __FILE__ , " $Revision$ " )
2005-06-06 22:39:32 +00:00
# include <sys/time.h>
# include <sys/signal.h>
2006-08-05 06:39:43 +00:00
# include <sys/stat.h>
2005-06-06 22:39:32 +00:00
# include <netinet/in.h>
2008-02-09 11:27:10 +00:00
# include "asterisk/paths.h" /* use ast_config_AST_DATA_DIR */
2005-04-21 06:02:45 +00:00
# include "asterisk/lock.h"
# include "asterisk/file.h"
# include "asterisk/channel.h"
# include "asterisk/pbx.h"
# include "asterisk/module.h"
# include "asterisk/translate.h"
# include "asterisk/say.h"
# include "asterisk/config.h"
# include "asterisk/features.h"
# include "asterisk/musiconhold.h"
# include "asterisk/callerid.h"
# include "asterisk/utils.h"
# include "asterisk/app.h"
# include "asterisk/causes.h"
2009-04-02 17:20:52 +00:00
# include "asterisk/rtp_engine.h"
Merged revisions 89622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line
closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27 06:47:08 +00:00
# include "asterisk/cdr.h"
2005-04-21 06:02:45 +00:00
# include "asterisk/manager.h"
2005-07-12 03:23:31 +00:00
# include "asterisk/privacy.h"
2006-02-01 23:05:28 +00:00
# include "asterisk/stringfields.h"
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
# include "asterisk/global_datastores.h"
2008-03-05 16:23:44 +00:00
# include "asterisk/dsp.h"
1999-12-04 21:35:07 +00:00
2008-11-01 21:10:07 +00:00
/*** DOCUMENTATION
< application name = " Dial " language = " en_US " >
< synopsis >
Attempt to connect to another device or endpoint and bridge the call .
< / synopsis >
< syntax >
< parameter name = " Technology/Resource " required = " true " argsep = " & " >
< argument name = " Technology/Resource " required = " true " >
< para > Specification of the device ( s ) to dial . These must be in the format of
< literal > Technology / Resource < / literal > , where < replaceable > Technology < / replaceable >
represents a particular channel driver , and < replaceable > Resource < / replaceable >
represents a resource available to that particular channel driver . < / para >
< / argument >
< argument name = " Technology2/Resource2 " required = " false " multiple = " true " >
< para > Optional extra devices to dial in parallel < / para >
< para > If you need more then one enter them as
Technology2 / Resource2 & amp ; Technology3 / Resourse3 & amp ; . . . . . < / para >
< / argument >
< / parameter >
< parameter name = " timeout " required = " false " >
< para > Specifies the number of seconds we attempt to dial the specified devices < / para >
< para > If not specified , this defaults to 136 years . < / para >
< / parameter >
< parameter name = " options " required = " false " >
< optionlist >
< option name = " A " >
< argument name = " x " required = " true " >
< para > The file to play to the called party < / para >
< / argument >
< para > Play an announcement to the called party , where < replaceable > x < / replaceable > is the prompt to be played < / para >
< / option >
< option name = " C " >
< para > Reset the call detail record ( CDR ) for this call . < / para >
< / option >
< option name = " c " >
< para > If the Dial ( ) application cancels this call , always set the flag to tell the channel
driver that the call is answered elsewhere . < / para >
< / option >
< option name = " d " >
< para > Allow the calling user to dial a 1 digit extension while waiting for
a call to be answered . Exit to that extension if it exists in the
current context , or the context defined in the < variable > EXITCONTEXT < / variable > variable ,
if it exists . < / para >
< / option >
< option name = " D " argsep = " : " >
< argument name = " called " / >
< argument name = " calling " / >
2009-03-17 17:17:51 +00:00
< argument name = " progress " / >
2008-11-01 21:10:07 +00:00
< para > Send the specified DTMF strings < emphasis > after < / emphasis > the called
party has answered , but before the call gets bridged . The
< replaceable > called < / replaceable > DTMF string is sent to the called party , and the
< replaceable > calling < / replaceable > DTMF string is sent to the calling party . Both arguments
2009-03-17 17:17:51 +00:00
can be used alone . If < replaceable > progress < / replaceable > is specified , its DTMF is sent
immediately after receiving a PROGRESS message . < / para >
2008-11-01 21:10:07 +00:00
< / option >
< option name = " e " >
< para > Execute the < literal > h < / literal > extension for peer after the call ends < / para >
< / option >
< option name = " f " >
< para > Force the callerid of the < emphasis > calling < / emphasis > channel to be set as the
extension associated with the channel using a dialplan < literal > hint < / literal > .
For example , some PSTNs do not allow CallerID to be set to anything
other than the number assigned to the caller . < / para >
< / option >
< option name = " F " argsep = " ^ " >
< argument name = " context " required = " false " / >
< argument name = " exten " required = " false " / >
< argument name = " priority " required = " true " / >
< para > When the caller hangs up , transfer the called party
to the specified destination and continue execution at that location . < / para >
< / option >
2009-04-09 19:10:02 +00:00
< option name = " F " >
< para > Proceed with dialplan execution at the next priority in the current extension if the
source channel hangs up . < / para >
< / option >
2008-11-01 21:10:07 +00:00
< option name = " g " >
< para > Proceed with dialplan execution at the next priority in the current extension if the
destination channel hangs up . < / para >
< / option >
< option name = " G " argsep = " ^ " >
< argument name = " context " required = " false " / >
< argument name = " exten " required = " false " / >
< argument name = " priority " required = " true " / >
< para > If the call is answered , transfer the calling party to
the specified < replaceable > priority < / replaceable > and the called party to the specified
< replaceable > priority < / replaceable > plus one . < / para >
< note >
< para > You cannot use any additional action post answer options in conjunction with this option . < / para >
< / note >
< / option >
< option name = " h " >
< para > Allow the called party to hang up by sending the < literal > * < / literal > DTMF digit . < / para >
< / option >
< option name = " H " >
< para > Allow the calling party to hang up by hitting the < literal > * < / literal > DTMF digit . < / para >
< / option >
< option name = " i " >
< para > Asterisk will ignore any forwarding requests it may receive on this dial attempt . < / para >
< / option >
2009-04-03 22:41:46 +00:00
< option name = " I " >
< para > Asterisk will ignore any connected line update requests or redirecting party update
requests it may receiveon this dial attempt . < / para >
< / option >
2008-11-01 21:10:07 +00:00
< option name = " k " >
< para > Allow the called party to enable parking of the call by sending
the DTMF sequence defined for call parking in < filename > features . conf < / filename > . < / para >
< / option >
< option name = " K " >
< para > Allow the calling party to enable parking of the call by sending
the DTMF sequence defined for call parking in < filename > features . conf < / filename > . < / para >
< / option >
< option name = " L " argsep = " : " >
< argument name = " x " required = " true " >
< para > Maximum call time , in milliseconds < / para >
< / argument >
< argument name = " y " >
< para > Warning time , in milliseconds < / para >
< / argument >
< argument name = " z " >
< para > Repeat time , in milliseconds < / para >
< / argument >
< para > Limit the call to < replaceable > x < / replaceable > milliseconds . Play a warning when < replaceable > y < / replaceable > milliseconds are
left . Repeat the warning every < replaceable > z < / replaceable > milliseconds until time expires . < / para >
< para > This option is affected by the following variables : < / para >
< variablelist >
< variable name = " LIMIT_PLAYAUDIO_CALLER " >
< value name = " yes " default = " true " / >
< value name = " no " / >
< para > If set , this variable causes Asterisk to play the prompts to the caller . < / para >
< / variable >
< variable name = " LIMIT_PLAYAUDIO_CALLEE " >
< value name = " yes " / >
< value name = " no " default = " true " / >
< para > If set , this variable causes Asterisk to play the prompts to the callee . < / para >
< / variable >
< variable name = " LIMIT_TIMEOUT_FILE " >
< value name = " filename " / >
< para > If specified , < replaceable > filename < / replaceable > specifies the sound prompt to play when the timeout is reached .
If not set , the time remaining will be announced . < / para >
< / variable >
< variable name = " LIMIT_CONNECT_FILE " >
< value name = " filename " / >
< para > If specified , < replaceable > filename < / replaceable > specifies the sound prompt to play when the call begins .
If not set , the time remaining will be announced . < / para >
< / variable >
< variable name = " LIMIT_WARNING_FILE " >
< value name = " filename " / >
< para > If specified , < replaceable > filename < / replaceable > specifies the sound prompt to play as
a warning when time < replaceable > x < / replaceable > is reached . If not set , the time remaining will be announced . < / para >
< / variable >
< / variablelist >
< / option >
< option name = " m " >
< argument name = " class " required = " false " / >
< para > Provide hold music to the calling party until a requested
channel answers . A specific music on hold < replaceable > class < / replaceable >
( as defined in < filename > musiconhold . conf < / filename > ) can be specified . < / para >
< / option >
< option name = " M " argsep = " ^ " >
< argument name = " macro " required = " true " >
< para > Name of the macro that should be executed . < / para >
< / argument >
< argument name = " arg " multiple = " true " >
< para > Macro arguments < / para >
< / argument >
< para > Execute the specified < replaceable > macro < / replaceable > for the < emphasis > called < / emphasis > channel
before connecting to the calling channel . Arguments can be specified to the Macro
using < literal > ^ < / literal > as a delimiter . The macro can set the variable
< variable > MACRO_RESULT < / variable > to specify the following actions after the macro is
finished executing : < / para >
< variablelist >
< variable name = " MACRO_RESULT " >
< para > If set , this action will be taken after the macro finished executing . < / para >
< value name = " ABORT " >
Hangup both legs of the call
< / value >
< value name = " CONGESTION " >
Behave as if line congestion was encountered
< / value >
< value name = " BUSY " >
Behave as if a busy signal was encountered
< / value >
< value name = " CONTINUE " >
Hangup the called party and allow the calling party to continue dialplan execution at the next priority
< / value >
< ! - - TODO : Fix this syntax up , once we ' ve figured out how to specify the GOTO syntax - - >
< value name = " GOTO:<context>^<exten>^<priority> " >
Transfer the call to the specified destination .
< / value >
< / variable >
< / variablelist >
< note >
< para > You cannot use any additional action post answer options in conjunction
with this option . Also , pbx services are not run on the peer ( called ) channel ,
so you will not be able to set timeouts via the TIMEOUT ( ) function in this macro . < / para >
< / note >
2009-05-18 14:45:23 +00:00
< warning > < para > Be aware of the limitations that macros have , specifically with regards to use of
the < literal > WaitExten < / literal > application . For more information , see the documentation for
Macro ( ) < / para > < / warning >
2008-11-01 21:10:07 +00:00
< / option >
< option name = " n " >
< para > This option is a modifier for the call screening / privacy mode . ( See the
< literal > p < / literal > and < literal > P < / literal > options . ) It specifies
that no introductions are to be saved in the < directory > priv - callerintros < / directory >
directory . < / para >
< / option >
< option name = " N " >
< para > This option is a modifier for the call screening / privacy mode . It specifies
that if Caller * ID is present , do not screen the call . < / para >
< / option >
< option name = " o " >
< para > Specify that the Caller * ID that was present on the < emphasis > calling < / emphasis > channel
be set as the Caller * ID on the < emphasis > called < / emphasis > channel . This was the
behavior of Asterisk 1.0 and earlier . < / para >
< / option >
< option name = " O " >
< argument name = " mode " >
< para > With < replaceable > mode < / replaceable > either not specified or set to < literal > 1 < / literal > ,
the originator hanging up will cause the phone to ring back immediately . < / para >
< para > With < replaceable > mode < / replaceable > set to < literal > 2 < / literal > , when the operator
flashes the trunk , it will ring their phone back . < / para >
< / argument >
< para > Enables < emphasis > operator services < / emphasis > mode . This option only
works when bridging a DAHDI channel to another DAHDI channel
only . if specified on non - DAHDI interfaces , it will be ignored .
When the destination answers ( presumably an operator services
station ) , the originator no longer has control of their line .
They may hang up , but the switch will not release their line
until the destination party ( the operator ) hangs up . < / para >
< / option >
< option name = " p " >
< para > This option enables screening mode . This is basically Privacy mode
without memory . < / para >
< / option >
< option name = " P " >
< argument name = " x " / >
< para > Enable privacy mode . Use < replaceable > x < / replaceable > as the family / key in the AstDB database if
it is provided . The current extension is used if a database family / key is not specified . < / para >
< / option >
< option name = " r " >
< para > Indicate ringing to the calling party , even if the called party isn ' t actually ringing . Pass no audio to the calling
party until the called channel has answered . < / para >
< / option >
< option name = " S " >
< argument name = " x " required = " true " / >
< para > Hang up the call < replaceable > x < / replaceable > seconds < emphasis > after < / emphasis > the called party has
answered the call . < / para >
< / option >
< option name = " t " >
< para > Allow the called party to transfer the calling party by sending the
DTMF sequence defined in < filename > features . conf < / filename > . < / para >
< / option >
< option name = " T " >
< para > Allow the calling party to transfer the called party by sending the
DTMF sequence defined in < filename > features . conf < / filename > . < / para >
< / option >
< option name = " U " argsep = " ^ " >
< argument name = " x " required = " true " >
< para > Name of the subroutine to execute via Gosub < / para >
< / argument >
< argument name = " arg " multiple = " true " required = " false " >
< para > Arguments for the Gosub routine < / para >
< / argument >
< para > Execute via Gosub the routine < replaceable > x < / replaceable > for the < emphasis > called < / emphasis > channel before connecting
to the calling channel . Arguments can be specified to the Gosub
using < literal > ^ < / literal > as a delimiter . The Gosub routine can set the variable
< variable > GOSUB_RESULT < / variable > to specify the following actions after the Gosub returns . < / para >
< variablelist >
< variable name = " GOSUB_RESULT " >
< value name = " ABORT " >
Hangup both legs of the call .
< / value >
< value name = " CONGESTION " >
Behave as if line congestion was encountered .
< / value >
< value name = " BUSY " >
Behave as if a busy signal was encountered .
< / value >
< value name = " CONTINUE " >
Hangup the called party and allow the calling party
to continue dialplan execution at the next priority .
< / value >
< ! - - TODO : Fix this syntax up , once we ' ve figured out how to specify the GOTO syntax - - >
< value name = " GOTO:<context>^<exten>^<priority> " >
Transfer the call to the specified priority . Optionally , an extension , or
extension and priority can be specified .
< / value >
< / variable >
< / variablelist >
< note >
< para > You cannot use any additional action post answer options in conjunction
with this option . Also , pbx services are not run on the peer ( called ) channel ,
so you will not be able to set timeouts via the TIMEOUT ( ) function in this routine . < / para >
< / note >
< / option >
< option name = " w " >
< para > Allow the called party to enable recording of the call by sending
the DTMF sequence defined for one - touch recording in < filename > features . conf < / filename > . < / para >
< / option >
< option name = " W " >
< para > Allow the calling party to enable recording of the call by sending
the DTMF sequence defined for one - touch recording in < filename > features . conf < / filename > . < / para >
< / option >
< option name = " x " >
< para > Allow the called party to enable recording of the call by sending
the DTMF sequence defined for one - touch automixmonitor in < filename > features . conf < / filename > . < / para >
< / option >
< option name = " X " >
< para > Allow the calling party to enable recording of the call by sending
the DTMF sequence defined for one - touch automixmonitor in < filename > features . conf < / filename > . < / para >
< / option >
2009-04-15 15:24:50 +00:00
< option name = " z " >
< para > On a call forward , cancel any dial timeout which has been set for this call . < / para >
< / option >
2008-11-01 21:10:07 +00:00
< / optionlist >
< / parameter >
< parameter name = " URL " >
< para > The optional URL will be sent to the called party if the channel driver supports it . < / para >
< / parameter >
< / syntax >
< description >
< para > This application will place calls to one or more specified channels . As soon
as one of the requested channels answers , the originating channel will be
answered , if it has not already been answered . These two channels will then
be active in a bridged call . All other channels that were requested will then
be hung up . < / para >
< para > Unless there is a timeout specified , the Dial application will wait
indefinitely until one of the called channels answers , the user hangs up , or
if all of the called channels are busy or unavailable . Dialplan executing will
continue if no requested channels can be called , or if the timeout expires .
This application will report normal termination if the originating channel
hangs up , or if the call is bridged and either of the parties in the bridge
ends the call . < / para >
< para > If the < variable > OUTBOUND_GROUP < / variable > variable is set , all peer channels created by this
application will be put into that group ( as in Set ( GROUP ( ) = . . . ) .
If the < variable > OUTBOUND_GROUP_ONCE < / variable > variable is set , all peer channels created by this
application will be put into that group ( as in Set ( GROUP ( ) = . . . ) . Unlike OUTBOUND_GROUP ,
however , the variable will be unset after use . < / para >
< para > This application sets the following channel variables : < / para >
< variablelist >
< variable name = " DIALEDTIME " >
< para > This is the time from dialing a channel until when it is disconnected . < / para >
< / variable >
< variable name = " ANSWEREDTIME " >
< para > This is the amount of time for actual call . < / para >
< / variable >
< variable name = " DIALSTATUS " >
< para > This is the status of the call < / para >
< value name = " CHANUNAVAIL " / >
< value name = " CONGESTION " / >
< value name = " NOANSWER " / >
< value name = " BUSY " / >
< value name = " ANSWER " / >
< value name = " CANCEL " / >
< value name = " DONTCALL " >
For the Privacy and Screening Modes .
Will be set if the called party chooses to send the calling party to the ' Go Away ' script .
< / value >
< value name = " TORTURE " >
For the Privacy and Screening Modes .
Will be set if the called party chooses to send the calling party to the ' torture ' script .
< / value >
< value name = " INVALIDARGS " / >
< / variable >
< / variablelist >
< / description >
< / application >
< application name = " RetryDial " language = " en_US " >
< synopsis >
Place a call , retrying on failure allowing an optional exit extension .
< / synopsis >
< syntax >
< parameter name = " announce " required = " true " >
< para > Filename of sound that will be played when no channel can be reached < / para >
< / parameter >
< parameter name = " sleep " required = " true " >
2008-11-02 02:50:33 +00:00
< para > Number of seconds to wait after a dial attempt failed before a new attempt is made < / para >
2008-11-01 21:10:07 +00:00
< / parameter >
< parameter name = " retries " required = " true " >
< para > Number of retries < / para >
< para > When this is reached flow will continue at the next priority in the dialplan < / para >
< / parameter >
< parameter name = " dialargs " required = " true " >
< para > Same format as arguments provided to the Dial application < / para >
< / parameter >
< / syntax >
< description >
< para > This application will attempt to place a call using the normal Dial application .
If no channel can be reached , the < replaceable > announce < / replaceable > file will be played .
Then , it will wait < replaceable > sleep < / replaceable > number of seconds before retrying the call .
After < replaceable > retries < / replaceable > number of attempts , the calling channel will continue at the next priority in the dialplan .
If the < replaceable > retries < / replaceable > setting is set to 0 , this application will retry endlessly .
While waiting to retry a call , a 1 digit extension may be dialed . If that
extension exists in either the context defined in < variable > EXITCONTEXT < / variable > or the current
one , The call will jump to that extension immediately .
The < replaceable > dialargs < / replaceable > are specified in the same format that arguments are provided
to the Dial application . < / para >
< / description >
< / application >
* * */
1999-12-04 21:35:07 +00:00
2008-11-01 21:10:07 +00:00
static char * app = " Dial " ;
2005-01-18 03:12:53 +00:00
static char * rapp = " RetryDial " ;
2005-11-03 21:40:36 +00:00
enum {
2007-12-12 20:05:13 +00:00
OPT_ANNOUNCE = ( 1 < < 0 ) ,
OPT_RESETCDR = ( 1 < < 1 ) ,
OPT_DTMF_EXIT = ( 1 < < 2 ) ,
OPT_SENDDTMF = ( 1 < < 3 ) ,
OPT_FORCECLID = ( 1 < < 4 ) ,
OPT_GO_ON = ( 1 < < 5 ) ,
OPT_CALLEE_HANGUP = ( 1 < < 6 ) ,
OPT_CALLER_HANGUP = ( 1 < < 7 ) ,
2009-04-03 22:41:46 +00:00
OPT_ORIGINAL_CLID = ( 1 < < 8 ) ,
2007-12-12 20:05:13 +00:00
OPT_DURATION_LIMIT = ( 1 < < 9 ) ,
OPT_MUSICBACK = ( 1 < < 10 ) ,
OPT_CALLEE_MACRO = ( 1 < < 11 ) ,
OPT_SCREEN_NOINTRO = ( 1 < < 12 ) ,
2009-04-03 22:41:46 +00:00
OPT_SCREEN_NOCALLERID = ( 1 < < 13 ) ,
OPT_IGNORE_CONNECTEDLINE = ( 1 < < 14 ) ,
2007-12-12 20:05:13 +00:00
OPT_SCREENING = ( 1 < < 15 ) ,
OPT_PRIVACY = ( 1 < < 16 ) ,
OPT_RINGBACK = ( 1 < < 17 ) ,
OPT_DURATION_STOP = ( 1 < < 18 ) ,
OPT_CALLEE_TRANSFER = ( 1 < < 19 ) ,
OPT_CALLER_TRANSFER = ( 1 < < 20 ) ,
OPT_CALLEE_MONITOR = ( 1 < < 21 ) ,
OPT_CALLER_MONITOR = ( 1 < < 22 ) ,
OPT_GOTO = ( 1 < < 23 ) ,
OPT_OPERMODE = ( 1 < < 24 ) ,
OPT_CALLEE_PARK = ( 1 < < 25 ) ,
OPT_CALLER_PARK = ( 1 < < 26 ) ,
2006-05-31 15:52:32 +00:00
OPT_IGNORE_FORWARDING = ( 1 < < 27 ) ,
2007-12-12 20:05:13 +00:00
OPT_CALLEE_GOSUB = ( 1 < < 28 ) ,
2007-11-30 21:19:57 +00:00
OPT_CALLEE_MIXMONITOR = ( 1 < < 29 ) ,
OPT_CALLER_MIXMONITOR = ( 1 < < 30 ) ,
2007-04-09 22:49:32 +00:00
} ;
2005-11-03 21:40:36 +00:00
2008-02-09 11:27:10 +00:00
# define DIAL_STILLGOING (1 << 31)
# define DIAL_NOFORWARDHTML ((uint64_t)1 << 32) /* flags are now 64 bits, so keep it up! */
2009-04-03 22:41:46 +00:00
# define DIAL_NOCONNECTEDLINE ((uint64_t)1 << 33)
# define OPT_CANCEL_ELSEWHERE ((uint64_t)1 << 34)
# define OPT_PEER_H ((uint64_t)1 << 35)
# define OPT_CALLEE_GO_ON ((uint64_t)1 << 36)
2009-04-15 15:24:50 +00:00
# define OPT_CANCEL_TIMEOUT ((uint64_t)1 << 37)
2005-11-03 21:40:36 +00:00
enum {
OPT_ARG_ANNOUNCE = 0 ,
OPT_ARG_SENDDTMF ,
OPT_ARG_GOTO ,
OPT_ARG_DURATION_LIMIT ,
OPT_ARG_MUSICBACK ,
OPT_ARG_CALLEE_MACRO ,
2007-06-19 23:36:34 +00:00
OPT_ARG_CALLEE_GOSUB ,
2008-04-09 13:55:28 +00:00
OPT_ARG_CALLEE_GO_ON ,
2005-11-03 21:40:36 +00:00
OPT_ARG_PRIVACY ,
OPT_ARG_DURATION_STOP ,
2006-04-22 11:30:06 +00:00
OPT_ARG_OPERMODE ,
2005-11-03 21:40:36 +00:00
/* note: this entry _MUST_ be the last one in the enum */
OPT_ARG_ARRAY_SIZE ,
2007-04-09 22:49:32 +00:00
} ;
2005-11-03 21:40:36 +00:00
2007-11-14 01:40:47 +00:00
AST_APP_OPTIONS ( dial_exec_options , BEGIN_OPTIONS
2005-11-03 21:40:36 +00:00
AST_APP_OPTION_ARG ( ' A ' , OPT_ANNOUNCE , OPT_ARG_ANNOUNCE ) ,
AST_APP_OPTION ( ' C ' , OPT_RESETCDR ) ,
2007-07-09 08:27:37 +00:00
AST_APP_OPTION ( ' c ' , OPT_CANCEL_ELSEWHERE ) ,
2005-11-03 21:40:36 +00:00
AST_APP_OPTION ( ' d ' , OPT_DTMF_EXIT ) ,
AST_APP_OPTION_ARG ( ' D ' , OPT_SENDDTMF , OPT_ARG_SENDDTMF ) ,
2007-07-17 19:40:29 +00:00
AST_APP_OPTION ( ' e ' , OPT_PEER_H ) ,
2005-11-03 21:40:36 +00:00
AST_APP_OPTION ( ' f ' , OPT_FORCECLID ) ,
2008-04-09 13:55:28 +00:00
AST_APP_OPTION_ARG ( ' F ' , OPT_CALLEE_GO_ON , OPT_ARG_CALLEE_GO_ON ) ,
2005-11-03 21:40:36 +00:00
AST_APP_OPTION ( ' g ' , OPT_GO_ON ) ,
AST_APP_OPTION_ARG ( ' G ' , OPT_GOTO , OPT_ARG_GOTO ) ,
AST_APP_OPTION ( ' h ' , OPT_CALLEE_HANGUP ) ,
AST_APP_OPTION ( ' H ' , OPT_CALLER_HANGUP ) ,
2006-05-31 15:52:32 +00:00
AST_APP_OPTION ( ' i ' , OPT_IGNORE_FORWARDING ) ,
2009-04-03 22:41:46 +00:00
AST_APP_OPTION ( ' I ' , OPT_IGNORE_CONNECTEDLINE ) ,
2007-06-19 23:36:34 +00:00
AST_APP_OPTION ( ' k ' , OPT_CALLEE_PARK ) ,
2007-08-31 18:46:02 +00:00
AST_APP_OPTION ( ' K ' , OPT_CALLER_PARK ) ,
2005-11-03 21:40:36 +00:00
AST_APP_OPTION_ARG ( ' L ' , OPT_DURATION_LIMIT , OPT_ARG_DURATION_LIMIT ) ,
AST_APP_OPTION_ARG ( ' m ' , OPT_MUSICBACK , OPT_ARG_MUSICBACK ) ,
AST_APP_OPTION_ARG ( ' M ' , OPT_CALLEE_MACRO , OPT_ARG_CALLEE_MACRO ) ,
AST_APP_OPTION ( ' n ' , OPT_SCREEN_NOINTRO ) ,
2009-04-03 22:41:46 +00:00
AST_APP_OPTION ( ' N ' , OPT_SCREEN_NOCALLERID ) ,
2005-11-03 21:40:36 +00:00
AST_APP_OPTION ( ' o ' , OPT_ORIGINAL_CLID ) ,
2007-12-12 20:05:13 +00:00
AST_APP_OPTION_ARG ( ' O ' , OPT_OPERMODE , OPT_ARG_OPERMODE ) ,
2005-11-03 21:40:36 +00:00
AST_APP_OPTION ( ' p ' , OPT_SCREENING ) ,
AST_APP_OPTION_ARG ( ' P ' , OPT_PRIVACY , OPT_ARG_PRIVACY ) ,
AST_APP_OPTION ( ' r ' , OPT_RINGBACK ) ,
AST_APP_OPTION_ARG ( ' S ' , OPT_DURATION_STOP , OPT_ARG_DURATION_STOP ) ,
AST_APP_OPTION ( ' t ' , OPT_CALLEE_TRANSFER ) ,
AST_APP_OPTION ( ' T ' , OPT_CALLER_TRANSFER ) ,
2007-06-19 23:36:34 +00:00
AST_APP_OPTION_ARG ( ' U ' , OPT_CALLEE_GOSUB , OPT_ARG_CALLEE_GOSUB ) ,
2005-11-03 21:40:36 +00:00
AST_APP_OPTION ( ' w ' , OPT_CALLEE_MONITOR ) ,
AST_APP_OPTION ( ' W ' , OPT_CALLER_MONITOR ) ,
2007-11-30 21:19:57 +00:00
AST_APP_OPTION ( ' x ' , OPT_CALLEE_MIXMONITOR ) ,
AST_APP_OPTION ( ' X ' , OPT_CALLER_MIXMONITOR ) ,
2009-04-15 15:24:50 +00:00
AST_APP_OPTION ( ' z ' , OPT_CANCEL_TIMEOUT ) ,
2007-11-14 01:40:47 +00:00
END_OPTIONS ) ;
2005-11-03 21:40:36 +00:00
2008-07-14 17:54:11 +00:00
# define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
2007-10-01 14:27:02 +00:00
OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
2008-07-14 17:54:11 +00:00
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK ) & & \
! chan - > audiohooks & & ! peer - > audiohooks )
2007-10-01 13:53:09 +00:00
2006-12-19 16:36:45 +00:00
/*
* The list of active channels
*/
struct chanlist {
struct chanlist * next ;
1999-12-04 21:35:07 +00:00
struct ast_channel * chan ;
2007-07-17 19:40:29 +00:00
uint64_t flags ;
2009-04-03 22:41:46 +00:00
struct ast_party_connected_line connected ;
1999-12-04 21:35:07 +00:00
} ;
Merged revisions 183126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-19 16:28:33 +00:00
static int detect_disconnect ( struct ast_channel * chan , char code , struct ast_str * featurecode ) ;
1999-12-04 21:35:07 +00:00
2007-07-09 08:27:37 +00:00
static void hanguptree ( struct chanlist * outgoing , struct ast_channel * exception , int answered_elsewhere )
1999-12-04 21:35:07 +00:00
{
/* Hang up a tree of stuff */
2006-12-19 16:36:45 +00:00
struct chanlist * oo ;
2005-01-18 03:12:53 +00:00
while ( outgoing ) {
1999-12-04 21:35:07 +00:00
/* Hangup any existing lines we have open */
2007-07-09 08:27:37 +00:00
if ( outgoing - > chan & & ( outgoing - > chan ! = exception ) ) {
2009-01-29 17:08:22 +00:00
if ( answered_elsewhere ) {
/* The flag is used for local channel inheritance and stuff */
2007-07-09 08:27:37 +00:00
ast_set_flag ( outgoing - > chan , AST_FLAG_ANSWERED_ELSEWHERE ) ;
2009-01-29 17:08:22 +00:00
/* This is for the channel drivers */
outgoing - > chan - > hangupcause = AST_CAUSE_ANSWERED_ELSEWHERE ;
}
2009-05-20 20:14:28 +00:00
ast_party_connected_line_free ( & outgoing - > connected ) ;
1999-12-04 21:35:07 +00:00
ast_hangup ( outgoing - > chan ) ;
2007-07-09 08:27:37 +00:00
}
1999-12-04 21:35:07 +00:00
oo = outgoing ;
2007-12-12 20:05:13 +00:00
outgoing = outgoing - > next ;
2007-06-06 21:20:11 +00:00
ast_free ( oo ) ;
1999-12-04 21:35:07 +00:00
}
}
2004-06-22 17:42:14 +00:00
# define AST_MAX_WATCHERS 256
2001-05-07 03:15:48 +00:00
2006-11-03 22:36:17 +00:00
/*
* argument to handle_cause ( ) and other functions .
*/
struct cause_args {
struct ast_channel * chan ;
int busy ;
int congestion ;
int nochan ;
} ;
static void handle_cause ( int cause , struct cause_args * num )
{
struct ast_cdr * cdr = num - > chan - > cdr ;
switch ( cause ) {
case AST_CAUSE_BUSY :
if ( cdr )
ast_cdr_busy ( cdr ) ;
num - > busy + + ;
break ;
case AST_CAUSE_CONGESTION :
if ( cdr )
ast_cdr_failed ( cdr ) ;
num - > congestion + + ;
break ;
2008-07-08 20:30:29 +00:00
case AST_CAUSE_NO_ROUTE_DESTINATION :
2006-11-03 22:36:17 +00:00
case AST_CAUSE_UNREGISTERED :
if ( cdr )
ast_cdr_failed ( cdr ) ;
num - > nochan + + ;
break ;
2009-04-20 21:24:34 +00:00
case AST_CAUSE_NO_ANSWER :
2009-04-20 21:29:29 +00:00
if ( cdr ) {
ast_cdr_noanswer ( cdr ) ;
}
break ;
2006-11-03 22:36:17 +00:00
case AST_CAUSE_NORMAL_CLEARING :
break ;
default :
num - > nochan + + ;
break ;
}
}
2005-01-18 03:12:53 +00:00
2006-11-03 21:51:16 +00:00
/* free the buffer if allocated, and set the pointer to the second arg */
# define S_REPLACE(s, new_val) \
do { \
if ( s ) \
2007-12-12 20:05:13 +00:00
ast_free ( s ) ; \
2006-11-03 21:51:16 +00:00
s = ( new_val ) ; \
} while ( 0 )
2005-01-18 03:12:53 +00:00
2008-02-09 11:27:10 +00:00
static int onedigit_goto ( struct ast_channel * chan , const char * context , char exten , int pri )
2005-01-18 03:12:53 +00:00
{
2005-04-29 15:04:26 +00:00
char rexten [ 2 ] = { exten , ' \0 ' } ;
2005-01-18 03:12:53 +00:00
if ( context ) {
2005-06-01 18:02:46 +00:00
if ( ! ast_goto_if_exists ( chan , context , rexten , pri ) )
2005-01-18 03:12:53 +00:00
return 1 ;
} else {
2005-06-01 18:02:46 +00:00
if ( ! ast_goto_if_exists ( chan , chan - > context , rexten , pri ) )
2005-01-18 03:12:53 +00:00
return 1 ;
2005-04-29 15:04:26 +00:00
else if ( ! ast_strlen_zero ( chan - > macrocontext ) ) {
2005-06-01 18:02:46 +00:00
if ( ! ast_goto_if_exists ( chan , chan - > macrocontext , rexten , pri ) )
2005-01-18 03:12:53 +00:00
return 1 ;
}
}
return 0 ;
}
2004-10-26 22:25:43 +00:00
2009-04-09 17:39:10 +00:00
/* do not call with chan lock held */
2006-04-19 14:14:40 +00:00
static const char * get_cid_name ( char * name , int namelen , struct ast_channel * chan )
2005-02-01 01:53:25 +00:00
{
2009-04-09 20:40:34 +00:00
const char * context ;
const char * exten ;
ast_channel_lock ( chan ) ;
context = ast_strdupa ( S_OR ( chan - > macrocontext , chan - > context ) ) ;
exten = ast_strdupa ( S_OR ( chan - > macroexten , chan - > exten ) ) ;
ast_channel_unlock ( chan ) ;
2005-02-01 01:53:25 +00:00
2006-04-19 14:14:40 +00:00
return ast_get_hint ( NULL , 0 , name , namelen , chan , context , exten ) ? name : " " ;
2005-02-01 01:53:25 +00:00
}
2007-12-06 15:04:34 +00:00
static void senddialevent ( struct ast_channel * src , struct ast_channel * dst , const char * dialstring )
2005-02-07 15:19:34 +00:00
{
2008-02-09 11:27:10 +00:00
manager_event ( EVENT_FLAG_CALL , " Dial " ,
" SubEvent: Begin \r \n "
" Channel: %s \r \n "
" Destination: %s \r \n "
" CallerIDNum: %s \r \n "
" CallerIDName: %s \r \n "
" UniqueID: %s \r \n "
" DestUniqueID: %s \r \n "
" Dialstring: %s \r \n " ,
src - > name , dst - > name , S_OR ( src - > cid . cid_num , " <unknown> " ) ,
S_OR ( src - > cid . cid_name , " <unknown> " ) , src - > uniqueid ,
dst - > uniqueid , dialstring ? dialstring : " " ) ;
2005-02-07 15:19:34 +00:00
}
2006-10-30 23:11:55 +00:00
static void senddialendevent ( const struct ast_channel * src , const char * dialstatus )
{
manager_event ( EVENT_FLAG_CALL , " Dial " ,
2008-02-09 11:27:10 +00:00
" SubEvent: End \r \n "
" Channel: %s \r \n "
" UniqueID: %s \r \n "
" DialStatus: %s \r \n " ,
src - > name , src - > uniqueid , dialstatus ) ;
}
2006-10-30 23:11:55 +00:00
2006-11-21 11:53:06 +00:00
/*!
* helper function for wait_for_answer ( )
*
* XXX this code is highly suspicious , as it essentially overwrites
* the outgoing channel without properly deleting it .
*/
2006-12-19 16:36:45 +00:00
static void do_forward ( struct chanlist * o ,
2009-04-15 15:24:50 +00:00
struct cause_args * num , struct ast_flags64 * peerflags , int single , int * to )
2006-11-04 00:01:40 +00:00
{
char tmpchan [ 256 ] ;
2007-02-27 22:17:42 +00:00
struct ast_channel * original = o - > chan ;
2006-11-04 00:01:40 +00:00
struct ast_channel * c = o - > chan ; /* the winner */
struct ast_channel * in = num - > chan ; /* the input channel */
2009-04-03 22:41:46 +00:00
struct ast_party_redirecting * apr = & o - > chan - > redirecting ;
struct ast_party_connected_line * apc = & o - > chan - > connected ;
2006-11-04 00:01:40 +00:00
char * stuff ;
char * tech ;
int cause ;
ast_copy_string ( tmpchan , c - > call_forward , sizeof ( tmpchan ) ) ;
if ( ( stuff = strchr ( tmpchan , ' / ' ) ) ) {
* stuff + + = ' \0 ' ;
tech = tmpchan ;
} else {
2008-06-18 13:09:02 +00:00
const char * forward_context ;
ast_channel_lock ( c ) ;
forward_context = pbx_builtin_getvar_helper ( c , " FORWARD_CONTEXT " ) ;
2006-11-04 00:01:40 +00:00
snprintf ( tmpchan , sizeof ( tmpchan ) , " %s@%s " , c - > call_forward , forward_context ? forward_context : c - > context ) ;
2008-06-18 13:09:02 +00:00
ast_channel_unlock ( c ) ;
2006-11-04 00:01:40 +00:00
stuff = tmpchan ;
tech = " Local " ;
}
/* Before processing channel, go ahead and check for forwarding */
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
ast_verb ( 3 , " Now forwarding %s to '%s/%s' (thanks to %s) \n " , in - > name , tech , stuff , c - > name ) ;
/* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
if ( ast_test_flag64 ( peerflags , OPT_IGNORE_FORWARDING ) ) {
ast_verb ( 3 , " Forwarding %s to '%s/%s' prevented. \n " , in - > name , tech , stuff ) ;
2006-11-04 00:01:40 +00:00
c = o - > chan = NULL ;
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
cause = AST_CAUSE_BUSY ;
} else {
/* Setup parameters */
c = o - > chan = ast_request ( tech , in - > nativeformats , stuff , & cause ) ;
if ( c ) {
if ( single )
ast_channel_make_compatible ( o - > chan , in ) ;
ast_channel_inherit_variables ( in , o - > chan ) ;
ast_channel_datastore_inherit ( in , o - > chan ) ;
} else
ast_log ( LOG_NOTICE , " Unable to create local channel for call forward to '%s/%s' (cause = %d) \n " , tech , stuff , cause ) ;
2006-11-04 00:01:40 +00:00
}
if ( ! c ) {
2008-02-09 11:27:10 +00:00
ast_clear_flag64 ( o , DIAL_STILLGOING ) ;
2006-11-04 00:01:40 +00:00
handle_cause ( cause , num ) ;
2008-11-26 19:57:11 +00:00
ast_hangup ( original ) ;
2006-11-04 00:01:40 +00:00
} else {
2009-04-02 17:20:52 +00:00
if ( single ) {
ast_rtp_instance_early_bridge_make_compatible ( c , in ) ;
}
2009-04-03 22:41:46 +00:00
c - > cdrflags = in - > cdrflags ;
ast_channel_set_redirecting ( c , apr ) ;
ast_channel_lock ( c ) ;
while ( ast_channel_trylock ( in ) ) {
CHANNEL_DEADLOCK_AVOIDANCE ( c ) ;
}
S_REPLACE ( c - > cid . cid_rdnis , ast_strdup ( S_OR ( original - > cid . cid_rdnis , S_OR ( in - > macroexten , in - > exten ) ) ) ) ;
c - > cid . cid_tns = in - > cid . cid_tns ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( o , OPT_FORCECLID ) ) {
2009-04-03 22:41:46 +00:00
S_REPLACE ( c - > cid . cid_num , ast_strdupa ( S_OR ( in - > macroexten , in - > exten ) ) ) ;
S_REPLACE ( c - > cid . cid_name , NULL ) ;
ast_string_field_set ( c , accountcode , c - > accountcode ) ;
2006-11-04 00:01:40 +00:00
} else {
2009-04-03 22:41:46 +00:00
ast_party_caller_copy ( & c - > cid , & in - > cid ) ;
ast_string_field_set ( c , accountcode , in - > accountcode ) ;
2006-11-04 00:01:40 +00:00
}
2009-04-03 22:41:46 +00:00
ast_party_connected_line_copy ( & c - > connected , apc ) ;
S_REPLACE ( in - > cid . cid_rdnis , ast_strdup ( c - > cid . cid_rdnis ) ) ;
ast_channel_update_redirecting ( in , apr ) ;
ast_clear_flag64 ( peerflags , OPT_IGNORE_CONNECTEDLINE ) ;
2009-04-15 15:24:50 +00:00
if ( ast_test_flag64 ( peerflags , OPT_CANCEL_TIMEOUT ) ) {
* to = - 1 ;
}
2006-11-04 00:01:40 +00:00
2009-04-10 15:49:16 +00:00
ast_channel_unlock ( in ) ;
ast_channel_unlock ( c ) ;
2006-11-04 00:01:40 +00:00
if ( ast_call ( c , tmpchan , 0 ) ) {
ast_log ( LOG_NOTICE , " Failed to dial on local channel for call forward to '%s' \n " , tmpchan ) ;
2008-02-09 11:27:10 +00:00
ast_clear_flag64 ( o , DIAL_STILLGOING ) ;
2007-02-27 22:17:42 +00:00
ast_hangup ( original ) ;
2008-11-26 19:57:11 +00:00
ast_hangup ( c ) ;
2006-11-04 00:01:40 +00:00
c = o - > chan = NULL ;
num - > nochan + + ;
} else {
2009-04-10 15:49:16 +00:00
ast_channel_lock ( c ) ;
while ( ast_channel_trylock ( in ) ) {
CHANNEL_DEADLOCK_AVOIDANCE ( c ) ;
}
2007-12-06 15:04:34 +00:00
senddialevent ( in , c , stuff ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ! ast_test_flag64 ( peerflags , OPT_ORIGINAL_CLID ) ) {
2007-06-04 18:00:24 +00:00
char cidname [ AST_MAX_EXTENSION ] = " " ;
2009-04-09 20:40:34 +00:00
const char * tmpexten ;
tmpexten = ast_strdupa ( S_OR ( in - > macroexten , in - > exten ) ) ;
ast_channel_unlock ( in ) ;
ast_channel_unlock ( c ) ;
ast_set_callerid ( c , tmpexten , get_cid_name ( cidname , sizeof ( cidname ) , in ) , NULL ) ;
} else {
ast_channel_unlock ( in ) ;
ast_channel_unlock ( c ) ;
2006-11-04 00:01:40 +00:00
}
2006-11-21 11:53:06 +00:00
/* Hangup the original channel now, in case we needed it */
2007-02-27 22:17:42 +00:00
ast_hangup ( original ) ;
2006-11-04 00:01:40 +00:00
}
2009-01-23 19:09:18 +00:00
if ( single ) {
ast_indicate ( in , - 1 ) ;
}
2006-11-04 00:01:40 +00:00
}
}
2006-11-04 11:00:49 +00:00
/* argument used for some functions. */
struct privacy_args {
2008-02-09 11:27:10 +00:00
int sentringing ;
int privdb_val ;
char privcid [ 256 ] ;
char privintro [ 1024 ] ;
char status [ 256 ] ;
2006-11-04 11:00:49 +00:00
} ;
static struct ast_channel * wait_for_answer ( struct ast_channel * in ,
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
struct chanlist * outgoing , int * to , struct ast_flags64 * peerflags ,
2006-11-04 11:00:49 +00:00
struct privacy_args * pa ,
2009-03-17 17:17:51 +00:00
const struct cause_args * num_in , int * result , char * dtmf_progress )
1999-12-04 21:35:07 +00:00
{
2006-11-03 22:36:17 +00:00
struct cause_args num = * num_in ;
int prestart = num . busy + num . congestion + num . nochan ;
1999-12-04 21:35:07 +00:00
int orig = * to ;
struct ast_channel * peer = NULL ;
2006-04-19 16:10:11 +00:00
/* single is set if only one destination is enabled */
2009-04-03 22:41:46 +00:00
int single = outgoing & & ! outgoing - > next ;
2007-08-08 21:44:58 +00:00
# ifdef HAVE_EPOLL
struct chanlist * epollo ;
# endif
2009-04-03 22:41:46 +00:00
struct ast_party_connected_line connected_caller ;
Merged revisions 183126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-19 16:28:33 +00:00
struct ast_str * featurecode = ast_str_alloca ( FEATURE_MAX_LEN + 1 ) ;
2009-05-05 20:54:07 +00:00
ast_party_connected_line_init ( & connected_caller ) ;
2001-05-07 03:15:48 +00:00
if ( single ) {
2003-11-05 20:53:49 +00:00
/* Turn off hold music, etc */
2009-04-03 22:41:46 +00:00
if ( ! ast_test_flag64 ( outgoing , OPT_MUSICBACK | OPT_RINGBACK ) )
ast_deactivate_generator ( in ) ;
2001-05-07 03:15:48 +00:00
/* If we are calling a single channel, make them compatible for in-band tone purpose */
ast_channel_make_compatible ( outgoing - > chan , in ) ;
2009-04-03 22:41:46 +00:00
if ( ! ast_test_flag64 ( peerflags , OPT_IGNORE_CONNECTEDLINE ) & & ! ast_test_flag64 ( outgoing , DIAL_NOCONNECTEDLINE ) ) {
ast_channel_lock ( outgoing - > chan ) ;
ast_connected_line_copy_from_caller ( & connected_caller , & outgoing - > chan - > cid ) ;
ast_channel_unlock ( outgoing - > chan ) ;
connected_caller . source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER ;
ast_channel_update_connected_line ( in , & connected_caller ) ;
ast_party_connected_line_free ( & connected_caller ) ;
}
2001-05-07 03:15:48 +00:00
}
2007-08-08 21:44:58 +00:00
# ifdef HAVE_EPOLL
for ( epollo = outgoing ; epollo ; epollo = epollo - > next )
ast_poll_channel_add ( in , epollo - > chan ) ;
2008-02-09 11:27:10 +00:00
# endif
2005-01-18 03:12:53 +00:00
while ( * to & & ! peer ) {
2006-12-19 16:36:45 +00:00
struct chanlist * o ;
2008-02-09 11:27:10 +00:00
int pos = 0 ; /* how many channels do we handle */
2006-04-19 14:02:49 +00:00
int numlines = prestart ;
2006-04-19 16:10:11 +00:00
struct ast_channel * winner ;
struct ast_channel * watchers [ AST_MAX_WATCHERS ] ;
2006-04-19 14:02:49 +00:00
watchers [ pos + + ] = in ;
for ( o = outgoing ; o ; o = o - > next ) {
2001-05-07 03:15:48 +00:00
/* Keep track of important channels */
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( o , DIAL_STILLGOING ) & & o - > chan )
2001-05-07 03:15:48 +00:00
watchers [ pos + + ] = o - > chan ;
numlines + + ;
1999-12-04 21:35:07 +00:00
}
2008-02-09 11:27:10 +00:00
if ( pos = = 1 ) { /* only the input channel is available */
2006-11-03 22:36:17 +00:00
if ( numlines = = ( num . busy + num . congestion + num . nochan ) ) {
2007-07-26 15:49:18 +00:00
ast_verb ( 2 , " Everyone is busy/congested at this time (%d:%d/%d/%d) \n " , numlines , num . busy , num . congestion , num . nochan ) ;
2006-11-03 22:36:17 +00:00
if ( num . busy )
2008-02-09 11:27:10 +00:00
strcpy ( pa - > status , " BUSY " ) ;
2006-11-03 22:36:17 +00:00
else if ( num . congestion )
2006-11-04 11:00:49 +00:00
strcpy ( pa - > status , " CONGESTION " ) ;
2006-11-03 22:36:17 +00:00
else if ( num . nochan )
2006-11-04 11:00:49 +00:00
strcpy ( pa - > status , " CHANUNAVAIL " ) ;
1999-12-04 21:35:07 +00:00
} else {
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " No one is available to answer at this time (%d:%d/%d/%d) \n " , numlines , num . busy , num . congestion , num . nochan ) ;
1999-12-04 21:35:07 +00:00
}
2001-05-07 03:15:48 +00:00
* to = 0 ;
1999-12-04 21:35:07 +00:00
return NULL ;
}
2001-05-07 03:15:48 +00:00
winner = ast_waitfor_n ( watchers , pos , to ) ;
2006-04-19 14:02:49 +00:00
for ( o = outgoing ; o ; o = o - > next ) {
struct ast_frame * f ;
struct ast_channel * c = o - > chan ;
if ( c = = NULL )
continue ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( o , DIAL_STILLGOING ) & & c - > _state = = AST_STATE_UP ) {
2002-09-02 15:20:28 +00:00
if ( ! peer ) {
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " %s answered %s \n " , c - > name , in - > name ) ;
2009-04-03 22:41:46 +00:00
if ( ! single & & ! ast_test_flag64 ( peerflags , OPT_IGNORE_CONNECTEDLINE ) ) {
if ( o - > connected . id . number ) {
ast_channel_update_connected_line ( in , & o - > connected ) ;
} else if ( ! ast_test_flag64 ( o , DIAL_NOCONNECTEDLINE ) ) {
ast_channel_lock ( c ) ;
ast_connected_line_copy_from_caller ( & connected_caller , & c - > cid ) ;
ast_channel_unlock ( c ) ;
connected_caller . source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER ;
ast_channel_update_connected_line ( in , & connected_caller ) ;
ast_party_connected_line_free ( & connected_caller ) ;
}
}
2006-04-19 14:02:49 +00:00
peer = c ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
ast_copy_flags64 ( peerflags , o ,
2008-02-09 11:27:10 +00:00
OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
OPT_CALLEE_PARK | OPT_CALLER_PARK |
OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
DIAL_NOFORWARDHTML ) ;
2008-07-01 16:16:36 +00:00
ast_string_field_set ( c , dialcontext , " " ) ;
2006-10-13 21:20:18 +00:00
ast_copy_string ( c - > exten , " " , sizeof ( c - > exten ) ) ;
2002-09-02 15:20:28 +00:00
}
2006-04-19 14:02:49 +00:00
continue ;
}
if ( c ! = winner )
continue ;
2006-11-04 00:01:40 +00:00
/* here, o->chan == c == winner */
2006-04-19 14:02:49 +00:00
if ( ! ast_strlen_zero ( c - > call_forward ) ) {
2009-04-15 15:24:50 +00:00
do_forward ( o , & num , peerflags , single , to ) ;
2006-04-19 14:53:18 +00:00
continue ;
2006-04-19 14:02:49 +00:00
}
f = ast_read ( winner ) ;
if ( ! f ) {
2006-04-19 14:53:18 +00:00
in - > hangupcause = c - > hangupcause ;
2007-12-03 14:14:43 +00:00
# ifdef HAVE_EPOLL
ast_poll_channel_del ( in , c ) ;
# endif
2006-04-19 14:53:18 +00:00
ast_hangup ( c ) ;
c = o - > chan = NULL ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
ast_clear_flag64 ( o , DIAL_STILLGOING ) ;
2006-11-03 22:36:17 +00:00
handle_cause ( in - > hangupcause , & num ) ;
2006-04-19 14:53:18 +00:00
continue ;
}
if ( f - > frametype = = AST_FRAME_CONTROL ) {
switch ( f - > subclass ) {
case AST_CONTROL_ANSWER :
/* This is our guy if someone answered. */
if ( ! peer ) {
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " %s answered %s \n " , c - > name , in - > name ) ;
2009-04-03 22:41:46 +00:00
if ( ! single & & ! ast_test_flag64 ( peerflags , OPT_IGNORE_CONNECTEDLINE ) ) {
if ( o - > connected . id . number ) {
ast_channel_update_connected_line ( in , & o - > connected ) ;
} else if ( ! ast_test_flag64 ( o , DIAL_NOCONNECTEDLINE ) ) {
ast_channel_lock ( c ) ;
ast_connected_line_copy_from_caller ( & connected_caller , & c - > cid ) ;
ast_channel_unlock ( c ) ;
connected_caller . source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER ;
ast_channel_update_connected_line ( in , & connected_caller ) ;
ast_party_connected_line_free ( & connected_caller ) ;
}
}
2006-04-19 14:53:18 +00:00
peer = c ;
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
if ( peer - > cdr ) {
peer - > cdr - > answer = ast_tvnow ( ) ;
peer - > cdr - > disposition = AST_CDR_ANSWERED ;
}
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
ast_copy_flags64 ( peerflags , o ,
2008-02-09 11:27:10 +00:00
OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
OPT_CALLEE_PARK | OPT_CALLER_PARK |
OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
DIAL_NOFORWARDHTML ) ;
2008-07-01 16:16:36 +00:00
ast_string_field_set ( c , dialcontext , " " ) ;
2006-10-13 21:20:18 +00:00
ast_copy_string ( c - > exten , " " , sizeof ( c - > exten ) ) ;
2008-07-14 17:54:11 +00:00
if ( CAN_EARLY_BRIDGE ( peerflags , in , peer ) )
2007-10-01 13:53:09 +00:00
/* Setup early bridge if appropriate */
ast_channel_early_bridge ( in , peer ) ;
2006-04-19 14:53:18 +00:00
}
/* If call has been answered, then the eventual hangup is likely to be normal hangup */
in - > hangupcause = AST_CAUSE_NORMAL_CLEARING ;
c - > hangupcause = AST_CAUSE_NORMAL_CLEARING ;
break ;
case AST_CONTROL_BUSY :
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " %s is busy \n " , c - > name ) ;
2006-04-19 14:53:18 +00:00
in - > hangupcause = c - > hangupcause ;
ast_hangup ( c ) ;
c = o - > chan = NULL ;
2008-02-09 11:27:10 +00:00
ast_clear_flag64 ( o , DIAL_STILLGOING ) ;
2006-11-03 22:36:17 +00:00
handle_cause ( AST_CAUSE_BUSY , & num ) ;
2006-04-19 14:53:18 +00:00
break ;
case AST_CONTROL_CONGESTION :
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " %s is circuit-busy \n " , c - > name ) ;
2006-04-19 14:50:17 +00:00
in - > hangupcause = c - > hangupcause ;
ast_hangup ( c ) ;
c = o - > chan = NULL ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
ast_clear_flag64 ( o , DIAL_STILLGOING ) ;
2006-11-03 22:36:17 +00:00
handle_cause ( AST_CAUSE_CONGESTION , & num ) ;
2006-04-19 14:53:18 +00:00
break ;
case AST_CONTROL_RINGING :
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " %s is ringing \n " , c - > name ) ;
2006-05-09 11:44:50 +00:00
/* Setup early media if appropriate */
2008-07-14 17:54:11 +00:00
if ( single & & CAN_EARLY_BRIDGE ( peerflags , in , c ) )
2006-09-21 19:27:26 +00:00
ast_channel_early_bridge ( in , c ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ! ( pa - > sentringing ) & & ! ast_test_flag64 ( outgoing , OPT_MUSICBACK ) ) {
2006-04-19 14:53:18 +00:00
ast_indicate ( in , AST_CONTROL_RINGING ) ;
2006-11-04 11:00:49 +00:00
pa - > sentringing + + ;
2006-04-19 14:53:18 +00:00
}
break ;
case AST_CONTROL_PROGRESS :
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " %s is making progress passing it to %s \n " , c - > name , in - > name ) ;
2006-05-09 11:44:50 +00:00
/* Setup early media if appropriate */
2008-07-14 17:54:11 +00:00
if ( single & & CAN_EARLY_BRIDGE ( peerflags , in , c ) )
2006-09-21 19:27:26 +00:00
ast_channel_early_bridge ( in , c ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ! ast_test_flag64 ( outgoing , OPT_RINGBACK ) )
2006-04-19 14:53:18 +00:00
ast_indicate ( in , AST_CONTROL_PROGRESS ) ;
2009-03-17 17:17:51 +00:00
if ( ! ast_strlen_zero ( dtmf_progress ) ) {
ast_verb ( 3 , " Sending DTMF '%s' to the called party as result of receiving a PROGRESS message. \n " , dtmf_progress ) ;
ast_dtmf_stream ( c , in , dtmf_progress , 250 , 0 ) ;
}
2006-04-19 14:53:18 +00:00
break ;
case AST_CONTROL_VIDUPDATE :
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " %s requested a video update, passing it to %s \n " , c - > name , in - > name ) ;
2006-04-19 14:53:18 +00:00
ast_indicate ( in , AST_CONTROL_VIDUPDATE ) ;
break ;
2008-03-05 22:43:22 +00:00
case AST_CONTROL_SRCUPDATE :
ast_verb ( 3 , " %s requested a source update, passing it to %s \n " , c - > name , in - > name ) ;
ast_indicate ( in , AST_CONTROL_SRCUPDATE ) ;
break ;
2009-04-03 22:41:46 +00:00
case AST_CONTROL_CONNECTED_LINE :
if ( ast_test_flag64 ( peerflags , OPT_IGNORE_CONNECTEDLINE ) ) {
ast_verb ( 3 , " Connected line update to %s prevented. \n " , in - > name ) ;
} else if ( ! single ) {
struct ast_party_connected_line connected ;
ast_verb ( 3 , " %s connected line has changed. Saving it until answer for %s \n " , c - > name , in - > name ) ;
ast_party_connected_line_set_init ( & connected , & o - > connected ) ;
ast_connected_line_parse_data ( f - > data . ptr , f - > datalen , & connected ) ;
ast_party_connected_line_set ( & o - > connected , & connected ) ;
ast_party_connected_line_free ( & connected ) ;
} else {
ast_verb ( 3 , " %s connected line has changed, passing it to %s \n " , c - > name , in - > name ) ;
ast_indicate_data ( in , AST_CONTROL_CONNECTED_LINE , f - > data . ptr , f - > datalen ) ;
}
break ;
case AST_CONTROL_REDIRECTING :
if ( ast_test_flag64 ( peerflags , OPT_IGNORE_CONNECTEDLINE ) ) {
ast_verb ( 3 , " Redirecting update to %s prevented. \n " , in - > name ) ;
} else {
ast_verb ( 3 , " %s redirecting info has changed, passing it to %s \n " , c - > name , in - > name ) ;
ast_indicate_data ( in , AST_CONTROL_REDIRECTING , f - > data . ptr , f - > datalen ) ;
}
break ;
2006-04-19 14:53:18 +00:00
case AST_CONTROL_PROCEEDING :
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " %s is proceeding passing it to %s \n " , c - > name , in - > name ) ;
2008-07-14 17:54:11 +00:00
if ( single & & CAN_EARLY_BRIDGE ( peerflags , in , c ) )
2006-09-21 19:27:26 +00:00
ast_channel_early_bridge ( in , c ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ! ast_test_flag64 ( outgoing , OPT_RINGBACK ) )
2006-04-19 14:53:18 +00:00
ast_indicate ( in , AST_CONTROL_PROCEEDING ) ;
break ;
case AST_CONTROL_HOLD :
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " Call on %s placed on hold \n " , c - > name ) ;
2006-04-19 14:53:18 +00:00
ast_indicate ( in , AST_CONTROL_HOLD ) ;
break ;
case AST_CONTROL_UNHOLD :
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " Call on %s left from hold \n " , c - > name ) ;
2006-04-19 14:53:18 +00:00
ast_indicate ( in , AST_CONTROL_UNHOLD ) ;
break ;
case AST_CONTROL_OFFHOOK :
case AST_CONTROL_FLASH :
/* Ignore going off hook and flash */
break ;
case - 1 :
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ! ast_test_flag64 ( outgoing , OPT_RINGBACK | OPT_MUSICBACK ) ) {
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " %s stopped sounds \n " , c - > name ) ;
2006-04-19 14:53:18 +00:00
ast_indicate ( in , - 1 ) ;
2006-11-04 11:00:49 +00:00
pa - > sentringing = 0 ;
2006-04-19 14:53:18 +00:00
}
break ;
default :
2007-06-14 19:39:12 +00:00
ast_debug ( 1 , " Dunno what to do with control type %d \n " , f - > subclass ) ;
2006-04-19 14:53:18 +00:00
}
2006-04-19 15:15:03 +00:00
} else if ( single ) {
2008-03-11 15:59:32 +00:00
switch ( f - > frametype ) {
case AST_FRAME_VOICE :
case AST_FRAME_IMAGE :
case AST_FRAME_TEXT :
if ( ast_write ( in , f ) ) {
ast_log ( LOG_WARNING , " Unable to write frame \n " ) ;
}
break ;
case AST_FRAME_HTML :
2008-05-22 16:29:54 +00:00
if ( ! ast_test_flag64 ( outgoing , DIAL_NOFORWARDHTML ) & & ast_channel_sendhtml ( in , f - > subclass , f - > data . ptr , f - > datalen ) = = - 1 ) {
2008-03-11 15:59:32 +00:00
ast_log ( LOG_WARNING , " Unable to send URL \n " ) ;
}
break ;
default :
break ;
2006-04-19 15:15:03 +00:00
}
1999-12-04 21:35:07 +00:00
}
2006-04-19 14:02:49 +00:00
ast_frfree ( f ) ;
} /* end for */
2001-05-07 03:15:48 +00:00
if ( winner = = in ) {
2006-04-19 14:02:49 +00:00
struct ast_frame * f = ast_read ( in ) ;
1999-12-04 21:35:07 +00:00
#if 0
if ( f & & ( f - > frametype ! = AST_FRAME_VOICE ) )
2005-09-07 19:13:00 +00:00
printf ( " Frame type: %d, %d \n " , f - > frametype , f - > subclass ) ;
2001-10-15 17:39:25 +00:00
else if ( ! f | | ( f - > frametype ! = AST_FRAME_VOICE ) )
printf ( " Hangup received on %s \n " , in - > name ) ;
1999-12-04 21:35:07 +00:00
# endif
2001-10-15 17:39:25 +00:00
if ( ! f | | ( ( f - > frametype = = AST_FRAME_CONTROL ) & & ( f - > subclass = = AST_CONTROL_HANGUP ) ) ) {
1999-12-04 21:35:07 +00:00
/* Got hung up */
2006-01-17 18:54:56 +00:00
* to = - 1 ;
2006-11-04 11:00:49 +00:00
strcpy ( pa - > status , " CANCEL " ) ;
Merged revisions 65200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines
Merged revisions 65172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 22:33:51 +00:00
ast_cdr_noanswer ( in - > cdr ) ;
2008-04-24 22:16:48 +00:00
if ( f ) {
2008-05-22 16:29:54 +00:00
if ( f - > data . uint32 ) {
in - > hangupcause = f - > data . uint32 ;
}
2004-12-06 17:12:21 +00:00
ast_frfree ( f ) ;
2008-04-24 22:16:48 +00:00
}
1999-12-04 21:35:07 +00:00
return NULL ;
}
2005-01-18 03:12:53 +00:00
2006-11-03 22:01:34 +00:00
/* now f is guaranteed non-NULL */
if ( f - > frametype = = AST_FRAME_DTMF ) {
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( peerflags , OPT_DTMF_EXIT ) ) {
2008-06-18 13:09:02 +00:00
const char * context ;
ast_channel_lock ( in ) ;
context = pbx_builtin_getvar_helper ( in , " EXITCONTEXT " ) ;
2005-04-29 15:04:26 +00:00
if ( onedigit_goto ( in , context , ( char ) f - > subclass , 1 ) ) {
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " User hit %c to disconnect call. \n " , f - > subclass ) ;
2007-12-12 20:05:13 +00:00
* to = 0 ;
Merged revisions 65200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines
Merged revisions 65172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 22:33:51 +00:00
ast_cdr_noanswer ( in - > cdr ) ;
2005-01-18 03:12:53 +00:00
* result = f - > subclass ;
2006-11-04 11:00:49 +00:00
strcpy ( pa - > status , " CANCEL " ) ;
2005-01-18 03:12:53 +00:00
ast_frfree ( f ) ;
2008-06-18 13:09:02 +00:00
ast_channel_unlock ( in ) ;
2005-01-18 03:12:53 +00:00
return NULL ;
}
2008-06-18 13:09:02 +00:00
ast_channel_unlock ( in ) ;
2005-01-18 03:12:53 +00:00
}
2008-02-09 11:27:10 +00:00
if ( ast_test_flag64 ( peerflags , OPT_CALLER_HANGUP ) & &
Merged revisions 183126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-19 16:28:33 +00:00
detect_disconnect ( in , f - > subclass , featurecode ) ) {
ast_verb ( 3 , " User requested call disconnect. \n " ) ;
2007-12-12 20:05:13 +00:00
* to = 0 ;
2006-11-04 11:00:49 +00:00
strcpy ( pa - > status , " CANCEL " ) ;
Merged revisions 65200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines
Merged revisions 65172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 22:33:51 +00:00
ast_cdr_noanswer ( in - > cdr ) ;
2005-01-18 03:12:53 +00:00
ast_frfree ( f ) ;
return NULL ;
}
2002-09-02 15:20:28 +00:00
}
2005-01-18 03:12:53 +00:00
2005-02-28 06:06:42 +00:00
/* Forward HTML stuff */
2008-02-09 11:27:10 +00:00
if ( single & & ( f - > frametype = = AST_FRAME_HTML ) & & ! ast_test_flag64 ( outgoing , DIAL_NOFORWARDHTML ) )
2008-05-22 16:29:54 +00:00
if ( ast_channel_sendhtml ( outgoing - > chan , f - > subclass , f - > data . ptr , f - > datalen ) = = - 1 )
2006-01-17 18:54:56 +00:00
ast_log ( LOG_WARNING , " Unable to send URL \n " ) ;
2005-02-28 06:06:42 +00:00
2007-02-14 21:10:53 +00:00
if ( single & & ( ( f - > frametype = = AST_FRAME_VOICE ) | | ( f - > frametype = = AST_FRAME_DTMF_BEGIN ) | | ( f - > frametype = = AST_FRAME_DTMF_END ) ) ) {
2002-09-02 15:20:28 +00:00
if ( ast_write ( outgoing - > chan , f ) )
2007-02-14 21:10:53 +00:00
ast_log ( LOG_WARNING , " Unable to forward voice or dtmf \n " ) ;
2002-09-02 15:20:28 +00:00
}
2008-02-09 11:27:10 +00:00
if ( single & & ( f - > frametype = = AST_FRAME_CONTROL ) & &
( ( f - > subclass = = AST_CONTROL_HOLD ) | |
( f - > subclass = = AST_CONTROL_UNHOLD ) | |
2008-03-05 22:43:22 +00:00
( f - > subclass = = AST_CONTROL_VIDUPDATE ) | |
2009-04-03 22:41:46 +00:00
( f - > subclass = = AST_CONTROL_SRCUPDATE ) | |
( f - > subclass = = AST_CONTROL_CONNECTED_LINE ) | |
( f - > subclass = = AST_CONTROL_REDIRECTING ) ) ) {
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " %s requested special control %d, passing it to %s \n " , in - > name , f - > subclass , outgoing - > chan - > name ) ;
2008-05-22 16:29:54 +00:00
ast_indicate_data ( outgoing - > chan , f - > subclass , f - > data . ptr , f - > datalen ) ;
2005-08-30 02:12:09 +00:00
}
2005-08-03 20:17:53 +00:00
ast_frfree ( f ) ;
1999-12-04 21:35:07 +00:00
}
2007-07-26 15:49:18 +00:00
if ( ! * to )
ast_verb ( 3 , " Nobody picked up in %d ms \n " , orig ) ;
2008-02-09 11:27:10 +00:00
if ( ! * to | | ast_check_hangup ( in ) )
Merged revisions 65200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines
Merged revisions 65172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 22:33:51 +00:00
ast_cdr_noanswer ( in - > cdr ) ;
1999-12-04 21:35:07 +00:00
}
2008-02-09 11:27:10 +00:00
2007-08-08 21:44:58 +00:00
# ifdef HAVE_EPOLL
2007-12-03 14:14:43 +00:00
for ( epollo = outgoing ; epollo ; epollo = epollo - > next ) {
if ( epollo - > chan )
ast_poll_channel_del ( in , epollo - > chan ) ;
}
2007-08-08 21:44:58 +00:00
# endif
1999-12-04 21:35:07 +00:00
return peer ;
2006-04-19 14:02:49 +00:00
}
Merged revisions 183126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-19 16:28:33 +00:00
static int detect_disconnect ( struct ast_channel * chan , char code , struct ast_str * featurecode )
{
struct ast_flags features = { AST_FEATURE_DISCONNECT } ; /* only concerned with disconnect feature */
2009-03-19 20:30:39 +00:00
struct ast_call_feature feature = { 0 , } ;
Merged revisions 183126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-19 16:28:33 +00:00
int res ;
ast_str_append ( & featurecode , 1 , " %c " , code ) ;
res = ast_feature_detect ( chan , & features , ast_str_buffer ( featurecode ) , & feature ) ;
if ( res ! = AST_FEATURE_RETURN_STOREDIGITS ) {
ast_str_reset ( featurecode ) ;
}
if ( feature . feature_mask & AST_FEATURE_DISCONNECT ) {
return 1 ;
}
return 0 ;
}
2006-04-19 14:02:49 +00:00
static void replace_macro_delimiter ( char * s )
{
for ( ; * s ; s + + )
if ( * s = = ' ^ ' )
2007-07-23 19:51:41 +00:00
* s = ' , ' ;
1999-12-04 21:35:07 +00:00
}
2006-04-19 17:58:07 +00:00
/* returns true if there is a valid privacy reply */
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
static int valid_priv_reply ( struct ast_flags64 * opts , int res )
2006-04-19 17:58:07 +00:00
{
if ( res < ' 1 ' )
return 0 ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( opts , OPT_PRIVACY ) & & res < = ' 5 ' )
2006-04-19 17:58:07 +00:00
return 1 ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( opts , OPT_SCREENING ) & & res < = ' 4 ' )
2006-04-19 17:58:07 +00:00
return 1 ;
return 0 ;
}
2006-11-04 00:50:18 +00:00
static int do_timelimit ( struct ast_channel * chan , struct ast_bridge_config * config ,
2008-11-12 21:34:51 +00:00
char * parse , struct timeval * calldurationlimit )
2006-11-04 00:50:18 +00:00
{
2006-11-14 20:09:10 +00:00
char * stringp = ast_strdupa ( parse ) ;
2006-11-04 00:50:18 +00:00
char * limit_str , * warning_str , * warnfreq_str ;
const char * var ;
2007-12-12 20:05:13 +00:00
int play_to_caller = 0 , play_to_callee = 0 ;
2006-11-04 00:50:18 +00:00
int delta ;
2006-11-14 20:09:10 +00:00
limit_str = strsep ( & stringp , " : " ) ;
warning_str = strsep ( & stringp , " : " ) ;
warnfreq_str = strsep ( & stringp , " : " ) ;
2006-11-04 00:50:18 +00:00
config - > timelimit = atol ( limit_str ) ;
if ( warning_str )
config - > play_warning = atol ( warning_str ) ;
if ( warnfreq_str )
config - > warning_freq = atol ( warnfreq_str ) ;
if ( ! config - > timelimit ) {
ast_log ( LOG_WARNING , " Dial does not accept L(%s), hanging up. \n " , limit_str ) ;
config - > timelimit = config - > play_warning = config - > warning_freq = 0 ;
config - > warning_sound = NULL ;
2008-02-09 11:27:10 +00:00
return - 1 ; /* error */
2006-11-04 00:50:18 +00:00
} else if ( ( delta = config - > play_warning - config - > timelimit ) > 0 ) {
int w = config - > warning_freq ;
/* If the first warning is requested _after_ the entire call would end,
and no warning frequency is requested , then turn off the warning . If
a warning frequency is requested , reduce the ' first warning ' time by
that frequency until it falls within the call ' s total time limit .
Graphically :
timelim - > | delta | < - playwarning
0 __________________ | _________________ |
| w | | | |
so the number of intervals to cut is 1 + ( delta - 1 ) / w
*/
if ( w = = 0 ) {
config - > play_warning = 0 ;
} else {
config - > play_warning - = w * ( 1 + ( delta - 1 ) / w ) ;
if ( config - > play_warning < 1 )
config - > play_warning = config - > warning_freq = 0 ;
}
}
2008-06-18 13:09:02 +00:00
ast_channel_lock ( chan ) ;
2006-11-04 00:50:18 +00:00
2007-12-12 20:05:13 +00:00
var = pbx_builtin_getvar_helper ( chan , " LIMIT_PLAYAUDIO_CALLER " ) ;
2008-06-18 13:09:02 +00:00
2006-11-04 00:50:18 +00:00
play_to_caller = var ? ast_true ( var ) : 1 ;
2008-02-09 11:27:10 +00:00
2007-12-12 20:05:13 +00:00
var = pbx_builtin_getvar_helper ( chan , " LIMIT_PLAYAUDIO_CALLEE " ) ;
2006-11-04 00:50:18 +00:00
play_to_callee = var ? ast_true ( var ) : 0 ;
2008-02-09 11:27:10 +00:00
2006-11-04 00:50:18 +00:00
if ( ! play_to_caller & & ! play_to_callee )
play_to_caller = 1 ;
2008-02-09 11:27:10 +00:00
2007-12-12 20:05:13 +00:00
var = pbx_builtin_getvar_helper ( chan , " LIMIT_WARNING_FILE " ) ;
2008-11-17 22:25:06 +00:00
config - > warning_sound = ! ast_strlen_zero ( var ) ? ast_strdup ( var ) : ast_strdup ( " timeleft " ) ;
2006-11-04 00:50:18 +00:00
/* The code looking at config wants a NULL, not just "", to decide
* that the message should not be played , so we replace " " with NULL .
* Note , pbx_builtin_getvar_helper _can_ return NULL if the variable is
* not found .
*/
2008-06-18 13:09:02 +00:00
2007-12-12 20:05:13 +00:00
var = pbx_builtin_getvar_helper ( chan , " LIMIT_TIMEOUT_FILE " ) ;
2008-11-17 22:25:06 +00:00
config - > end_sound = ! ast_strlen_zero ( var ) ? ast_strdup ( var ) : NULL ;
2008-06-18 13:09:02 +00:00
2007-12-12 20:05:13 +00:00
var = pbx_builtin_getvar_helper ( chan , " LIMIT_CONNECT_FILE " ) ;
2008-11-17 22:25:06 +00:00
config - > start_sound = ! ast_strlen_zero ( var ) ? ast_strdup ( var ) : NULL ;
2008-06-18 13:09:02 +00:00
ast_channel_unlock ( chan ) ;
2006-11-04 00:50:18 +00:00
/* undo effect of S(x) in case they are both used */
2008-11-12 21:34:51 +00:00
calldurationlimit - > tv_sec = 0 ;
calldurationlimit - > tv_usec = 0 ;
2006-11-04 00:50:18 +00:00
/* more efficient to do it like S(x) does since no advanced opts */
if ( ! config - > play_warning & & ! config - > start_sound & & ! config - > end_sound & & config - > timelimit ) {
2008-11-12 21:34:51 +00:00
calldurationlimit - > tv_sec = config - > timelimit / 1000 ;
calldurationlimit - > tv_usec = ( config - > timelimit % 1000 ) * 1000 ;
ast_verb ( 3 , " Setting call duration limit to %.3lf seconds. \n " ,
calldurationlimit - > tv_sec + calldurationlimit - > tv_usec / 1000000.0 ) ;
2006-11-04 00:50:18 +00:00
config - > timelimit = play_to_caller = play_to_callee =
config - > play_warning = config - > warning_freq = 0 ;
2007-07-26 15:49:18 +00:00
} else {
ast_verb ( 3 , " Limit Data for this call: \n " ) ;
ast_verb ( 4 , " timelimit = %ld \n " , config - > timelimit ) ;
ast_verb ( 4 , " play_warning = %ld \n " , config - > play_warning ) ;
ast_verb ( 4 , " play_to_caller = %s \n " , play_to_caller ? " yes " : " no " ) ;
ast_verb ( 4 , " play_to_callee = %s \n " , play_to_callee ? " yes " : " no " ) ;
ast_verb ( 4 , " warning_freq = %ld \n " , config - > warning_freq ) ;
ast_verb ( 4 , " start_sound = %s \n " , S_OR ( config - > start_sound , " " ) ) ;
ast_verb ( 4 , " warning_sound = %s \n " , config - > warning_sound ) ;
ast_verb ( 4 , " end_sound = %s \n " , S_OR ( config - > end_sound , " " ) ) ;
2006-11-04 00:50:18 +00:00
}
2008-02-09 11:27:10 +00:00
if ( play_to_caller )
ast_set_flag ( & ( config - > features_caller ) , AST_FEATURE_PLAY_WARNING ) ;
if ( play_to_callee )
ast_set_flag ( & ( config - > features_callee ) , AST_FEATURE_PLAY_WARNING ) ;
2006-11-04 00:50:18 +00:00
return 0 ;
}
2006-11-04 11:00:49 +00:00
static int do_privacy ( struct ast_channel * chan , struct ast_channel * peer ,
2008-02-09 11:27:10 +00:00
struct ast_flags64 * opts , char * * opt_args , struct privacy_args * pa )
2006-11-04 11:00:49 +00:00
{
int res2 ;
int loopcount = 0 ;
2008-02-09 11:27:10 +00:00
/* Get the user's intro, store it in priv-callerintros/$CID,
unless it is already there - - this should be done before the
2006-11-04 11:00:49 +00:00
call is actually dialed */
2008-02-09 11:27:10 +00:00
/* all ring indications and moh for the caller has been halted as soon as the
2006-11-04 11:00:49 +00:00
target extension was picked up . We are going to have to kill some
time and make the caller believe the peer hasn ' t picked up yet */
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( opts , OPT_MUSICBACK ) & & ! ast_strlen_zero ( opt_args [ OPT_ARG_MUSICBACK ] ) ) {
2007-02-09 19:39:26 +00:00
char * original_moh = ast_strdupa ( chan - > musicclass ) ;
2006-11-04 11:00:49 +00:00
ast_indicate ( chan , - 1 ) ;
2007-02-09 19:39:26 +00:00
ast_string_field_set ( chan , musicclass , opt_args [ OPT_ARG_MUSICBACK ] ) ;
2006-11-04 11:00:49 +00:00
ast_moh_start ( chan , opt_args [ OPT_ARG_MUSICBACK ] , NULL ) ;
2007-02-09 19:39:26 +00:00
ast_string_field_set ( chan , musicclass , original_moh ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
} else if ( ast_test_flag64 ( opts , OPT_RINGBACK ) ) {
2006-11-04 11:00:49 +00:00
ast_indicate ( chan , AST_CONTROL_RINGING ) ;
pa - > sentringing + + ;
}
/* Start autoservice on the other chan ?? */
res2 = ast_autoservice_start ( chan ) ;
/* Now Stream the File */
for ( loopcount = 0 ; loopcount < 3 ; loopcount + + ) {
2008-02-09 11:27:10 +00:00
if ( res2 & & loopcount = = 0 ) /* error in ast_autoservice_start() */
2006-11-04 11:00:49 +00:00
break ;
2008-02-09 11:27:10 +00:00
if ( ! res2 ) /* on timeout, play the message again */
2007-12-12 20:05:13 +00:00
res2 = ast_play_and_wait ( peer , " priv-callpending " ) ;
2006-11-04 11:00:49 +00:00
if ( ! valid_priv_reply ( opts , res2 ) )
res2 = 0 ;
2008-02-09 11:27:10 +00:00
/* priv-callpending script:
2006-11-04 11:00:49 +00:00
" I have a caller waiting, who introduces themselves as: "
*/
if ( ! res2 )
res2 = ast_play_and_wait ( peer , pa - > privintro ) ;
if ( ! valid_priv_reply ( opts , res2 ) )
res2 = 0 ;
/* now get input from the called party, as to their choice */
2007-12-12 20:05:13 +00:00
if ( ! res2 ) {
2006-11-04 11:00:49 +00:00
/* XXX can we have both, or they are mutually exclusive ? */
2007-12-12 20:05:13 +00:00
if ( ast_test_flag64 ( opts , OPT_PRIVACY ) )
res2 = ast_play_and_wait ( peer , " priv-callee-options " ) ;
if ( ast_test_flag64 ( opts , OPT_SCREENING ) )
res2 = ast_play_and_wait ( peer , " screen-callee-options " ) ;
2006-11-04 11:00:49 +00:00
}
/*! \page DialPrivacy Dial Privacy scripts
\ par priv - callee - options script :
" Dial 1 if you wish this caller to reach you directly in the future,
and immediately connect to their incoming call
2008-02-09 11:27:10 +00:00
Dial 2 if you wish to send this caller to voicemail now and
2006-11-04 11:00:49 +00:00
forevermore .
Dial 3 to send this caller to the torture menus , now and forevermore .
Dial 4 to send this caller to a simple " go away " menu , now and forevermore .
Dial 5 to allow this caller to come straight thru to you in the future ,
but right now , just this once , send them to voicemail . "
\ par screen - callee - options script :
" Dial 1 if you wish to immediately connect to the incoming call
Dial 2 if you wish to send this caller to voicemail .
Dial 3 to send this caller to the torture menus .
Dial 4 to send this caller to a simple " go away " menu .
*/
if ( valid_priv_reply ( opts , res2 ) )
break ;
/* invalid option */
res2 = ast_play_and_wait ( peer , " vm-sorry " ) ;
}
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( opts , OPT_MUSICBACK ) ) {
2006-11-04 11:00:49 +00:00
ast_moh_stop ( chan ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
} else if ( ast_test_flag64 ( opts , OPT_RINGBACK ) ) {
2006-11-04 11:00:49 +00:00
ast_indicate ( chan , - 1 ) ;
2007-12-12 20:05:13 +00:00
pa - > sentringing = 0 ;
2006-11-04 11:00:49 +00:00
}
ast_autoservice_stop ( chan ) ;
2007-12-12 20:05:13 +00:00
if ( ast_test_flag64 ( opts , OPT_PRIVACY ) & & ( res2 > = ' 1 ' & & res2 < = ' 5 ' ) ) {
2006-11-04 11:00:49 +00:00
/* map keypresses to various things, the index is res2 - '1' */
2009-05-12 13:59:35 +00:00
static const char * const _val [ ] = { " ALLOW " , " DENY " , " TORTURE " , " KILL " , " ALLOW " } ;
2006-11-04 11:00:49 +00:00
static const int _flag [ ] = { AST_PRIVACY_ALLOW , AST_PRIVACY_DENY , AST_PRIVACY_TORTURE , AST_PRIVACY_KILL , AST_PRIVACY_ALLOW } ;
int i = res2 - ' 1 ' ;
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " --Set privacy database entry %s/%s to %s \n " ,
2008-02-09 11:27:10 +00:00
opt_args [ OPT_ARG_PRIVACY ] , pa - > privcid , _val [ i ] ) ;
2006-11-04 11:00:49 +00:00
ast_privacy_set ( opt_args [ OPT_ARG_PRIVACY ] , pa - > privcid , _flag [ i ] ) ;
}
switch ( res2 ) {
case ' 1 ' :
break ;
case ' 2 ' :
ast_copy_string ( pa - > status , " NOANSWER " , sizeof ( pa - > status ) ) ;
break ;
case ' 3 ' :
ast_copy_string ( pa - > status , " TORTURE " , sizeof ( pa - > status ) ) ;
break ;
case ' 4 ' :
ast_copy_string ( pa - > status , " DONTCALL " , sizeof ( pa - > status ) ) ;
break ;
case ' 5 ' :
/* XXX should we set status to DENY ? */
2007-12-12 20:05:13 +00:00
if ( ast_test_flag64 ( opts , OPT_PRIVACY ) )
2006-11-04 11:00:49 +00:00
break ;
/* if not privacy, then 5 is the same as "default" case */
2008-02-09 11:27:10 +00:00
default : /* bad input or -1 if failure to start autoservice */
2006-11-04 11:00:49 +00:00
/* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do? */
/* well, there seems basically two choices. Just patch the caller thru immediately,
or , . . . put ' em thru to voicemail . */
/* since the callee may have hung up, let's do the voicemail thing, no database decision */
ast_log ( LOG_NOTICE , " privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding \n " ) ;
/* XXX should we set status to DENY ? */
/* XXX what about the privacy flags ? */
break ;
}
2008-02-09 11:27:10 +00:00
if ( res2 = = ' 1 ' ) { /* the only case where we actually connect */
/* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
2006-11-04 11:00:49 +00:00
just clog things up , and it ' s not useful information , not being tied to a CID */
2007-12-12 20:05:13 +00:00
if ( strncmp ( pa - > privcid , " NOCALLERID " , 10 ) = = 0 | | ast_test_flag64 ( opts , OPT_SCREEN_NOINTRO ) ) {
2006-11-04 11:00:49 +00:00
ast_filedelete ( pa - > privintro , NULL ) ;
2007-12-12 20:05:13 +00:00
if ( ast_fileexists ( pa - > privintro , NULL , NULL ) > 0 )
2006-11-04 11:00:49 +00:00
ast_log ( LOG_NOTICE , " privacy: ast_filedelete didn't do its job on %s \n " , pa - > privintro ) ;
2007-07-26 15:49:18 +00:00
else
ast_verb ( 3 , " Successfully deleted %s intro file \n " , pa - > privintro ) ;
2006-11-04 11:00:49 +00:00
}
2008-02-09 11:27:10 +00:00
return 0 ; /* the good exit path */
2006-11-04 11:00:49 +00:00
} else {
ast_hangup ( peer ) ; /* hang up on the callee -- he didn't want to talk anyway! */
return - 1 ;
}
}
2006-12-19 09:15:23 +00:00
/*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
static int setup_privacy_args ( struct privacy_args * pa ,
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
struct ast_flags64 * opts , char * opt_args [ ] , struct ast_channel * chan )
2006-12-19 09:15:23 +00:00
{
char callerid [ 60 ] ;
int res ;
char * l ;
2008-03-05 16:23:44 +00:00
int silencethreshold ;
2006-12-19 09:15:23 +00:00
if ( ! ast_strlen_zero ( chan - > cid . cid_num ) ) {
l = ast_strdupa ( chan - > cid . cid_num ) ;
ast_shrink_phone_number ( l ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( opts , OPT_PRIVACY ) ) {
2007-12-14 14:48:38 +00:00
ast_verb ( 3 , " Privacy DB is '%s', clid is '%s' \n " , opt_args [ OPT_ARG_PRIVACY ] , l ) ;
2006-12-19 09:15:23 +00:00
pa - > privdb_val = ast_privacy_check ( opt_args [ OPT_ARG_PRIVACY ] , l ) ;
} else {
2007-12-14 14:48:38 +00:00
ast_verb ( 3 , " Privacy Screening, clid is '%s' \n " , l ) ;
2006-12-19 09:15:23 +00:00
pa - > privdb_val = AST_PRIVACY_UNKNOWN ;
}
} else {
char * tnam , * tn2 ;
tnam = ast_strdupa ( chan - > name ) ;
/* clean the channel name so slashes don't try to end up in disk file name */
for ( tn2 = tnam ; * tn2 ; tn2 + + ) {
2007-12-14 14:48:38 +00:00
if ( * tn2 = = ' / ' ) /* any other chars to be afraid of? */
2006-12-19 09:15:23 +00:00
* tn2 = ' = ' ;
}
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " Privacy-- callerid is empty \n " ) ;
2006-12-19 09:15:23 +00:00
snprintf ( callerid , sizeof ( callerid ) , " NOCALLERID_%s%s " , chan - > exten , tnam ) ;
l = callerid ;
pa - > privdb_val = AST_PRIVACY_UNKNOWN ;
}
2008-02-09 11:27:10 +00:00
2007-12-12 20:05:13 +00:00
ast_copy_string ( pa - > privcid , l , sizeof ( pa - > privcid ) ) ;
2006-12-19 09:15:23 +00:00
2009-04-03 22:41:46 +00:00
if ( strncmp ( pa - > privcid , " NOCALLERID " , 10 ) ! = 0 & & ast_test_flag64 ( opts , OPT_SCREEN_NOCALLERID ) ) {
/* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
2007-12-14 14:48:38 +00:00
ast_verb ( 3 , " CallerID set (%s); N option set; Screening should be off \n " , pa - > privcid ) ;
2006-12-19 09:15:23 +00:00
pa - > privdb_val = AST_PRIVACY_ALLOW ;
2009-04-03 22:41:46 +00:00
} else if ( ast_test_flag64 ( opts , OPT_SCREEN_NOCALLERID ) & & strncmp ( pa - > privcid , " NOCALLERID " , 10 ) = = 0 ) {
2007-12-14 14:48:38 +00:00
ast_verb ( 3 , " CallerID blank; N option set; Screening should happen; dbval is %d \n " , pa - > privdb_val ) ;
2006-12-19 09:15:23 +00:00
}
2007-12-12 20:05:13 +00:00
if ( pa - > privdb_val = = AST_PRIVACY_DENY ) {
2007-12-14 14:48:38 +00:00
ast_verb ( 3 , " Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable \n " ) ;
2006-12-19 09:15:23 +00:00
ast_copy_string ( pa - > status , " NOANSWER " , sizeof ( pa - > status ) ) ;
return 0 ;
2007-12-12 20:05:13 +00:00
} else if ( pa - > privdb_val = = AST_PRIVACY_KILL ) {
2006-12-19 09:15:23 +00:00
ast_copy_string ( pa - > status , " DONTCALL " , sizeof ( pa - > status ) ) ;
return 0 ; /* Is this right? */
2007-12-12 20:05:13 +00:00
} else if ( pa - > privdb_val = = AST_PRIVACY_TORTURE ) {
2006-12-19 09:15:23 +00:00
ast_copy_string ( pa - > status , " TORTURE " , sizeof ( pa - > status ) ) ;
return 0 ; /* is this right??? */
2007-12-12 20:05:13 +00:00
} else if ( pa - > privdb_val = = AST_PRIVACY_UNKNOWN ) {
2008-02-09 11:27:10 +00:00
/* Get the user's intro, store it in priv-callerintros/$CID,
unless it is already there - - this should be done before the
2006-12-19 09:15:23 +00:00
call is actually dialed */
/* make sure the priv-callerintros dir actually exists */
snprintf ( pa - > privintro , sizeof ( pa - > privintro ) , " %s/sounds/priv-callerintros " , ast_config_AST_DATA_DIR ) ;
2007-06-22 04:35:12 +00:00
if ( ( res = ast_mkdir ( pa - > privintro , 0755 ) ) ) {
ast_log ( LOG_WARNING , " privacy: can't create directory priv-callerintros: %s \n " , strerror ( res ) ) ;
2006-12-19 09:15:23 +00:00
return - 1 ;
}
2007-06-22 04:35:12 +00:00
snprintf ( pa - > privintro , sizeof ( pa - > privintro ) , " priv-callerintros/%s " , pa - > privcid ) ;
if ( ast_fileexists ( pa - > privintro , NULL , NULL ) > 0 & & strncmp ( pa - > privcid , " NOCALLERID " , 10 ) ! = 0 ) {
2006-12-19 09:15:23 +00:00
/* the DELUX version of this code would allow this caller the
option to hear and retape their previously recorded intro .
*/
} else {
int duration ; /* for feedback from play_and_wait */
/* the file doesn't exist yet. Let the caller submit his
vocal intro for posterity */
/* priv-recordintro script:
" At the tone, please say your name: "
*/
2008-03-05 16:23:44 +00:00
silencethreshold = ast_dsp_get_threshold_from_settings ( THRESHOLD_SILENCE ) ;
2007-02-17 03:57:23 +00:00
ast_answer ( chan ) ;
2008-03-05 16:23:44 +00:00
res = ast_play_and_record ( chan , " priv-recordintro " , pa - > privintro , 4 , " gsm " , & duration , silencethreshold , 2000 , 0 ) ; /* NOTE: I've reduced the total time to 4 sec */
2006-12-19 09:15:23 +00:00
/* don't think we'll need a lock removed, we took care of
conflicts by naming the pa . privintro file */
if ( res = = - 1 ) {
/* Delete the file regardless since they hung up during recording */
ast_filedelete ( pa - > privintro , NULL ) ;
2007-12-12 20:05:13 +00:00
if ( ast_fileexists ( pa - > privintro , NULL , NULL ) > 0 )
ast_log ( LOG_NOTICE , " privacy: ast_filedelete didn't do its job on %s \n " , pa - > privintro ) ;
2007-07-26 15:49:18 +00:00
else
ast_verb ( 3 , " Successfully deleted %s intro file \n " , pa - > privintro ) ;
2006-12-19 09:15:23 +00:00
return - 1 ;
}
if ( ! ast_streamfile ( chan , " vm-dialout " , chan - > language ) )
ast_waitstream ( chan , " " ) ;
}
}
2008-02-09 11:27:10 +00:00
return 1 ; /* success */
2006-12-19 09:15:23 +00:00
}
2008-11-09 01:27:00 +00:00
static void end_bridge_callback ( void * data )
{
char buf [ 80 ] ;
time_t end ;
struct ast_channel * chan = data ;
2008-11-12 17:41:56 +00:00
if ( ! chan - > cdr ) {
return ;
}
2008-11-09 01:27:00 +00:00
time ( & end ) ;
ast_channel_lock ( chan ) ;
if ( chan - > cdr - > answer . tv_sec ) {
snprintf ( buf , sizeof ( buf ) , " %ld " , end - chan - > cdr - > answer . tv_sec ) ;
pbx_builtin_setvar_helper ( chan , " ANSWEREDTIME " , buf ) ;
}
if ( chan - > cdr - > start . tv_sec ) {
snprintf ( buf , sizeof ( buf ) , " %ld " , end - chan - > cdr - > start . tv_sec ) ;
pbx_builtin_setvar_helper ( chan , " DIALEDTIME " , buf ) ;
}
ast_channel_unlock ( chan ) ;
}
2008-11-18 18:31:08 +00:00
static void end_bridge_callback_data_fixup ( struct ast_bridge_config * bconfig , struct ast_channel * originator , struct ast_channel * terminator ) {
bconfig - > end_bridge_callback_data = originator ;
}
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
static int dial_exec_full ( struct ast_channel * chan , void * data , struct ast_flags64 * peerflags , int * continue_exec )
1999-12-04 21:35:07 +00:00
{
2008-02-09 11:27:10 +00:00
int res = - 1 ; /* default: error */
char * rest , * cur ; /* scan the list of destinations */
struct chanlist * outgoing = NULL ; /* list of destinations */
2004-06-21 13:30:58 +00:00
struct ast_channel * peer ;
2008-02-09 11:27:10 +00:00
int to ; /* timeout */
2006-11-03 22:36:17 +00:00
struct cause_args num = { chan , 0 , 0 , 0 } ;
2004-10-26 22:25:43 +00:00
int cause ;
2007-05-17 16:49:50 +00:00
char numsubst [ 256 ] ;
2007-06-04 18:00:24 +00:00
char cidname [ AST_MAX_EXTENSION ] = " " ;
2006-11-04 00:50:18 +00:00
2007-12-12 20:05:13 +00:00
struct ast_bridge_config config = { { 0 , } } ;
2008-11-12 21:34:51 +00:00
struct timeval calldurationlimit = { 0 , } ;
2009-03-17 17:17:51 +00:00
char * dtmfcalled = NULL , * dtmfcalling = NULL , * dtmf_progress = NULL ;
2006-11-04 11:00:49 +00:00
struct privacy_args pa = {
. sentringing = 0 ,
. privdb_val = 0 ,
2007-02-03 20:46:36 +00:00
. status = " INVALIDARGS " ,
2006-11-04 11:00:49 +00:00
} ;
2006-04-11 16:15:11 +00:00
int sentringing = 0 , moh = 0 ;
2005-12-20 17:52:31 +00:00
const char * outbound_group = NULL ;
2006-04-19 16:54:04 +00:00
int result = 0 ;
2005-11-02 21:46:52 +00:00
char * parse ;
2006-04-22 11:30:06 +00:00
int opermode = 0 ;
2005-11-02 21:46:52 +00:00
AST_DECLARE_APP_ARGS ( args ,
2008-02-09 11:27:10 +00:00
AST_APP_ARG ( peers ) ;
AST_APP_ARG ( timeout ) ;
AST_APP_ARG ( options ) ;
AST_APP_ARG ( url ) ;
2005-11-02 21:46:52 +00:00
) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
struct ast_flags64 opts = { 0 , } ;
2005-11-02 21:46:52 +00:00
char * opt_args [ OPT_ARG_ARRAY_SIZE ] ;
2007-12-04 17:35:40 +00:00
struct ast_datastore * datastore = NULL ;
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
int fulldial = 0 , num_dialed = 0 ;
2005-03-17 22:39:04 +00:00
2008-10-28 17:07:39 +00:00
/* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
pbx_builtin_setvar_helper ( chan , " DIALSTATUS " , " " ) ;
pbx_builtin_setvar_helper ( chan , " DIALEDPEERNUMBER " , " " ) ;
pbx_builtin_setvar_helper ( chan , " DIALEDPEERNAME " , " " ) ;
pbx_builtin_setvar_helper ( chan , " ANSWEREDTIME " , " " ) ;
pbx_builtin_setvar_helper ( chan , " DIALEDTIME " , " " ) ;
2005-10-26 19:48:14 +00:00
if ( ast_strlen_zero ( data ) ) {
2005-11-02 21:46:52 +00:00
ast_log ( LOG_WARNING , " Dial requires an argument (technology/number) \n " ) ;
2007-02-03 20:46:36 +00:00
pbx_builtin_setvar_helper ( chan , " DIALSTATUS " , pa . status ) ;
1999-12-04 21:35:07 +00:00
return - 1 ;
}
2004-09-22 05:19:06 +00:00
2006-05-10 13:22:15 +00:00
parse = ast_strdupa ( data ) ;
2008-02-09 11:27:10 +00:00
2005-11-02 21:46:52 +00:00
AST_STANDARD_APP_ARGS ( args , parse ) ;
2006-04-19 14:02:49 +00:00
if ( ! ast_strlen_zero ( args . options ) & &
2008-02-09 11:27:10 +00:00
ast_app_parse_options64 ( dial_exec_options , & opts , opt_args , args . options ) ) {
2007-02-03 20:46:36 +00:00
pbx_builtin_setvar_helper ( chan , " DIALSTATUS " , pa . status ) ;
2006-04-19 14:02:49 +00:00
goto done ;
2007-02-03 20:46:36 +00:00
}
2003-01-17 05:10:52 +00:00
2005-11-02 21:46:52 +00:00
if ( ast_strlen_zero ( args . peers ) ) {
ast_log ( LOG_WARNING , " Dial requires an argument (technology/number) \n " ) ;
2007-02-03 20:46:36 +00:00
pbx_builtin_setvar_helper ( chan , " DIALSTATUS " , pa . status ) ;
2006-04-19 14:02:49 +00:00
goto done ;
2005-11-02 21:46:52 +00:00
}
2004-04-26 23:22:34 +00:00
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( & opts , OPT_OPERMODE ) ) {
2006-11-04 01:16:20 +00:00
opermode = ast_strlen_zero ( opt_args [ OPT_ARG_OPERMODE ] ) ? 1 : atoi ( opt_args [ OPT_ARG_OPERMODE ] ) ;
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " Setting operator services mode to %d. \n " , opermode ) ;
2006-04-22 11:30:06 +00:00
}
2009-03-17 17:17:51 +00:00
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( & opts , OPT_DURATION_STOP ) & & ! ast_strlen_zero ( opt_args [ OPT_ARG_DURATION_STOP ] ) ) {
2008-11-12 21:34:51 +00:00
calldurationlimit . tv_sec = atoi ( opt_args [ OPT_ARG_DURATION_STOP ] ) ;
if ( ! calldurationlimit . tv_sec ) {
2006-04-11 16:15:11 +00:00
ast_log ( LOG_WARNING , " Dial does not accept S(%s), hanging up. \n " , opt_args [ OPT_ARG_DURATION_STOP ] ) ;
2007-02-03 20:46:36 +00:00
pbx_builtin_setvar_helper ( chan , " DIALSTATUS " , pa . status ) ;
2006-04-19 14:02:49 +00:00
goto done ;
2006-04-11 16:15:11 +00:00
}
2008-11-12 21:34:51 +00:00
ast_verb ( 3 , " Setting call duration limit to %.3lf seconds. \n " , calldurationlimit . tv_sec + calldurationlimit . tv_usec / 1000000.0 ) ;
2005-11-02 21:46:52 +00:00
}
2005-03-17 22:39:04 +00:00
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( & opts , OPT_SENDDTMF ) & & ! ast_strlen_zero ( opt_args [ OPT_ARG_SENDDTMF ] ) ) {
2009-03-17 17:17:51 +00:00
dtmf_progress = opt_args [ OPT_ARG_SENDDTMF ] ;
dtmfcalled = strsep ( & dtmf_progress , " : " ) ;
dtmfcalling = strsep ( & dtmf_progress , " : " ) ;
2005-11-02 21:46:52 +00:00
}
2005-03-17 22:39:04 +00:00
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( & opts , OPT_DURATION_LIMIT ) & & ! ast_strlen_zero ( opt_args [ OPT_ARG_DURATION_LIMIT ] ) ) {
2006-11-04 00:50:18 +00:00
if ( do_timelimit ( chan , & config , opt_args [ OPT_ARG_DURATION_LIMIT ] , & calldurationlimit ) )
2006-04-19 14:02:49 +00:00
goto done ;
2003-01-17 05:10:52 +00:00
}
2005-11-02 21:46:52 +00:00
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( & opts , OPT_RESETCDR ) & & chan - > cdr )
2005-11-06 21:00:35 +00:00
ast_cdr_reset ( chan - > cdr , NULL ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( & opts , OPT_PRIVACY ) & & ast_strlen_zero ( opt_args [ OPT_ARG_PRIVACY ] ) )
2005-11-02 21:46:52 +00:00
opt_args [ OPT_ARG_PRIVACY ] = ast_strdupa ( chan - > exten ) ;
2005-07-12 03:23:31 +00:00
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( & opts , OPT_PRIVACY ) | | ast_test_flag64 ( & opts , OPT_SCREENING ) ) {
2006-12-19 09:15:23 +00:00
res = setup_privacy_args ( & pa , & opts , opt_args , chan ) ;
if ( res < = 0 )
2005-07-12 03:23:31 +00:00
goto out ;
2008-02-09 11:27:10 +00:00
res = - 1 ; /* reset default */
2003-01-17 05:10:52 +00:00
}
2004-11-07 21:49:43 +00:00
Merged revisions 183126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-19 16:28:33 +00:00
if ( ast_test_flag64 ( & opts , OPT_DTMF_EXIT ) | | ast_test_flag64 ( & opts , OPT_CALLER_HANGUP ) ) {
2009-02-11 22:41:01 +00:00
__ast_answer ( chan , 0 , 0 ) ;
}
2007-02-15 16:24:13 +00:00
if ( continue_exec )
* continue_exec = 0 ;
2008-02-09 11:27:10 +00:00
2004-11-07 21:49:43 +00:00
/* If a channel group has been specified, get it for use when we create peer channels */
2008-06-18 13:09:02 +00:00
ast_channel_lock ( chan ) ;
2007-04-13 19:18:46 +00:00
if ( ( outbound_group = pbx_builtin_getvar_helper ( chan , " OUTBOUND_GROUP_ONCE " ) ) ) {
2008-06-18 13:09:02 +00:00
outbound_group = ast_strdupa ( outbound_group ) ;
2007-04-13 19:18:46 +00:00
pbx_builtin_setvar_helper ( chan , " OUTBOUND_GROUP_ONCE " , NULL ) ;
2008-06-18 13:09:02 +00:00
} else if ( ( outbound_group = pbx_builtin_getvar_helper ( chan , " OUTBOUND_GROUP " ) ) ) {
outbound_group = ast_strdupa ( outbound_group ) ;
2007-04-13 19:18:46 +00:00
}
2008-06-18 13:09:02 +00:00
ast_channel_unlock ( chan ) ;
2009-04-15 15:24:50 +00:00
ast_copy_flags64 ( peerflags , & opts , OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_IGNORE_CONNECTEDLINE | OPT_CANCEL_TIMEOUT ) ;
2008-03-01 01:30:37 +00:00
2006-04-19 14:02:49 +00:00
/* loop through the list of dial destinations */
rest = args . peers ;
while ( ( cur = strsep ( & rest , " & " ) ) ) {
2006-12-19 16:36:45 +00:00
struct chanlist * tmp ;
2008-02-09 11:27:10 +00:00
struct ast_channel * tc ; /* channel for this destination */
1999-12-18 07:01:48 +00:00
/* Get a technology/[device:]number pair */
2006-04-19 16:36:15 +00:00
char * number = cur ;
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
char * interface = ast_strdupa ( number ) ;
2006-04-19 16:36:15 +00:00
char * tech = strsep ( & number , " / " ) ;
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
/* find if we already dialed this interface */
struct ast_dialed_interface * di ;
AST_LIST_HEAD ( , ast_dialed_interface ) * dialed_interfaces ;
num_dialed + + ;
1999-12-04 21:35:07 +00:00
if ( ! number ) {
2005-11-02 21:46:52 +00:00
ast_log ( LOG_WARNING , " Dial argument takes format (technology/[device:]number1) \n " ) ;
1999-12-04 21:35:07 +00:00
goto out ;
}
2006-04-19 14:02:49 +00:00
if ( ! ( tmp = ast_calloc ( 1 , sizeof ( * tmp ) ) ) )
1999-12-04 21:35:07 +00:00
goto out ;
2005-11-02 21:46:52 +00:00
if ( opts . flags ) {
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
ast_copy_flags64 ( tmp , & opts ,
2008-02-09 11:27:10 +00:00
OPT_CANCEL_ELSEWHERE |
OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
OPT_CALLEE_PARK | OPT_CALLER_PARK |
OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID ) ;
ast_set2_flag64 ( tmp , args . url , DIAL_NOFORWARDHTML ) ;
2002-05-17 14:33:10 +00:00
}
2005-07-07 23:32:37 +00:00
ast_copy_string ( numsubst , number , sizeof ( numsubst ) ) ;
1999-12-04 21:35:07 +00:00
/* Request the peer */
2007-12-07 16:40:41 +00:00
ast_channel_lock ( chan ) ;
datastore = ast_channel_datastore_find ( chan , & dialed_interface_info , NULL ) ;
2009-04-03 22:41:46 +00:00
/* If the incoming channel has previously had connected line information
* set on it ( perhaps through the CONNECTED_LINE dialplan function ) then
* seed the calllist ' s connected line information with this previously
* acquired info
*/
if ( chan - > connected . id . number ) {
ast_party_connected_line_copy ( & tmp - > connected , & chan - > connected ) ;
}
2007-12-07 16:40:41 +00:00
ast_channel_unlock ( chan ) ;
if ( datastore )
dialed_interfaces = datastore - > data ;
else {
2008-08-05 16:56:11 +00:00
if ( ! ( datastore = ast_datastore_alloc ( & dialed_interface_info , NULL ) ) ) {
2008-02-09 11:27:10 +00:00
ast_log ( LOG_WARNING , " Unable to create channel datastore for dialed interfaces. Aborting! \n " ) ;
2007-12-12 20:05:13 +00:00
ast_free ( tmp ) ;
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
goto out ;
}
2007-12-07 16:40:41 +00:00
datastore - > inheritance = DATASTORE_INHERIT_FOREVER ;
if ( ! ( dialed_interfaces = ast_calloc ( 1 , sizeof ( * dialed_interfaces ) ) ) ) {
2007-12-12 20:05:13 +00:00
ast_free ( tmp ) ;
2007-12-07 16:40:41 +00:00
goto out ;
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
}
2007-12-07 16:40:41 +00:00
datastore - > data = dialed_interfaces ;
AST_LIST_HEAD_INIT ( dialed_interfaces ) ;
ast_channel_lock ( chan ) ;
ast_channel_datastore_add ( chan , datastore ) ;
ast_channel_unlock ( chan ) ;
}
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
AST_LIST_LOCK ( dialed_interfaces ) ;
AST_LIST_TRAVERSE ( dialed_interfaces , di , list ) {
2007-12-07 16:40:41 +00:00
if ( ! strcasecmp ( di - > interface , interface ) ) {
2008-02-09 11:27:10 +00:00
ast_log ( LOG_WARNING , " Skipping dialing interface '%s' again since it has already been dialed \n " ,
2007-12-07 16:40:41 +00:00
di - > interface ) ;
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
break ;
}
}
2007-12-07 16:40:41 +00:00
AST_LIST_UNLOCK ( dialed_interfaces ) ;
if ( di ) {
fulldial + + ;
2007-12-12 20:05:13 +00:00
ast_free ( tmp ) ;
2007-12-07 16:40:41 +00:00
continue ;
}
/* It is always ok to dial a Local interface. We only keep track of
* which " real " interfaces have been dialed . The Local channel will
* inherit this list so that if it ends up dialing a real interface ,
* it won ' t call one that has already been called . */
if ( strcasecmp ( tech , " Local " ) ) {
if ( ! ( di = ast_calloc ( 1 , sizeof ( * di ) + strlen ( interface ) ) ) ) {
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
AST_LIST_UNLOCK ( dialed_interfaces ) ;
2007-12-12 20:05:13 +00:00
ast_free ( tmp ) ;
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
goto out ;
}
strcpy ( di - > interface , interface ) ;
2007-12-07 16:40:41 +00:00
AST_LIST_LOCK ( dialed_interfaces ) ;
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
AST_LIST_INSERT_TAIL ( dialed_interfaces , di , list ) ;
2007-12-07 02:52:38 +00:00
AST_LIST_UNLOCK ( dialed_interfaces ) ;
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
}
2006-12-19 09:58:40 +00:00
tc = ast_request ( tech , chan - > nativeformats , numsubst , & cause ) ;
if ( ! tc ) {
1999-12-04 21:35:07 +00:00
/* If we can't, just go on to the next call */
2006-12-19 09:58:40 +00:00
ast_log ( LOG_WARNING , " Unable to create channel of type '%s' (cause %d - %s) \n " ,
2008-02-09 11:27:10 +00:00
tech , cause , ast_cause2str ( cause ) ) ;
2006-11-03 22:36:17 +00:00
handle_cause ( cause , & num ) ;
2008-02-09 11:27:10 +00:00
if ( ! rest ) /* we are on the last destination */
2005-10-05 21:18:28 +00:00
chan - > hangupcause = cause ;
2007-06-06 21:20:11 +00:00
ast_free ( tmp ) ;
1999-12-04 21:35:07 +00:00
continue ;
}
2006-12-19 09:58:40 +00:00
pbx_builtin_setvar_helper ( tc , " DIALEDPEERNUMBER " , numsubst ) ;
2004-11-01 02:23:28 +00:00
2009-04-03 22:41:46 +00:00
ast_channel_lock ( tc ) ;
while ( ast_channel_trylock ( chan ) ) {
CHANNEL_DEADLOCK_AVOIDANCE ( tc ) ;
}
2005-12-20 17:52:31 +00:00
/* Setup outgoing SDP to match incoming one */
2009-04-02 17:20:52 +00:00
if ( ! outgoing & & ! rest ) {
ast_rtp_instance_early_bridge_make_compatible ( tc , chan ) ;
}
2005-12-20 17:52:31 +00:00
2005-01-08 17:23:29 +00:00
/* Inherit specially named variables from parent channel */
2006-12-19 09:58:40 +00:00
ast_channel_inherit_variables ( chan , tc ) ;
2008-09-13 13:54:15 +00:00
ast_channel_datastore_inherit ( chan , tc ) ;
2004-11-01 02:23:28 +00:00
2006-12-19 09:58:40 +00:00
tc - > appl = " AppDial " ;
tc - > data = " (Outgoing Line) " ;
2008-05-01 23:06:23 +00:00
memset ( & tc - > whentohangup , 0 , sizeof ( tc - > whentohangup ) ) ;
2006-04-19 16:19:52 +00:00
2009-04-03 22:41:46 +00:00
/* If the new channel has no callerid, try to guess what it should be */
if ( ast_strlen_zero ( tc - > cid . cid_num ) ) {
if ( ! ast_strlen_zero ( chan - > connected . id . number ) ) {
ast_set_callerid ( tc , chan - > connected . id . number , chan - > connected . id . name , chan - > connected . ani ) ;
} else if ( ! ast_strlen_zero ( chan - > cid . cid_dnid ) ) {
ast_set_callerid ( tc , chan - > cid . cid_dnid , NULL , NULL ) ;
} else if ( ! ast_strlen_zero ( S_OR ( chan - > macroexten , chan - > exten ) ) ) {
ast_set_callerid ( tc , S_OR ( chan - > macroexten , chan - > exten ) , NULL , NULL ) ;
}
ast_set_flag64 ( tmp , DIAL_NOCONNECTEDLINE ) ;
}
2004-10-02 00:58:31 +00:00
2009-04-03 22:41:46 +00:00
ast_connected_line_copy_from_caller ( & tc - > connected , & chan - > cid ) ;
S_REPLACE ( tc - > cid . cid_rdnis , ast_strdup ( chan - > cid . cid_rdnis ) ) ;
ast_party_redirecting_copy ( & tc - > redirecting , & chan - > redirecting ) ;
tc - > cid . cid_tns = chan - > cid . cid_tns ;
2006-12-19 09:58:40 +00:00
ast_string_field_set ( tc , accountcode , chan - > accountcode ) ;
tc - > cdrflags = chan - > cdrflags ;
if ( ast_strlen_zero ( tc - > musicclass ) )
ast_string_field_set ( tc , musicclass , chan - > musicclass ) ;
2009-04-03 22:41:46 +00:00
/* Pass ADSI CPE and transfer capability */
2006-12-19 09:58:40 +00:00
tc - > adsicpe = chan - > adsicpe ;
tc - > transfercapability = chan - > transfercapability ;
2004-11-07 21:49:43 +00:00
/* If we have an outbound group, set this peer channel to it */
if ( outbound_group )
2006-12-19 09:58:40 +00:00
ast_app_group_set_channel ( tc , outbound_group ) ;
2009-01-29 17:08:22 +00:00
/* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
if ( ast_test_flag ( chan , AST_FLAG_ANSWERED_ELSEWHERE ) )
ast_set_flag ( tc , AST_FLAG_ANSWERED_ELSEWHERE ) ;
/* Check if we're forced by configuration */
if ( ast_test_flag64 ( & opts , OPT_CANCEL_ELSEWHERE ) )
ast_set_flag ( tc , AST_FLAG_ANSWERED_ELSEWHERE ) ;
2004-11-07 21:49:43 +00:00
2006-10-13 21:20:18 +00:00
/* Inherit context and extension */
2008-07-01 16:16:36 +00:00
ast_string_field_set ( tc , dialcontext , ast_strlen_zero ( chan - > macrocontext ) ? chan - > context : chan - > macrocontext ) ;
2007-02-16 18:53:17 +00:00
if ( ! ast_strlen_zero ( chan - > macroexten ) )
ast_copy_string ( tc - > exten , chan - > macroexten , sizeof ( tc - > exten ) ) ;
else
ast_copy_string ( tc - > exten , chan - > exten , sizeof ( tc - > exten ) ) ;
2006-10-13 21:20:18 +00:00
2009-04-10 17:32:25 +00:00
ast_channel_unlock ( tc ) ;
2008-02-09 11:27:10 +00:00
res = ast_call ( tc , numsubst , 0 ) ; /* Place the call, but don't wait on the answer */
2003-08-14 20:48:44 +00:00
/* Save the info in cdr's that we called them */
if ( chan - > cdr )
2006-12-19 09:58:40 +00:00
ast_cdr_setdestchan ( chan - > cdr , tc - > name ) ;
2003-08-14 20:48:44 +00:00
2004-06-20 06:24:25 +00:00
/* check the results of ast_call */
1999-12-04 21:35:07 +00:00
if ( res ) {
/* Again, keep going even if there's an error */
2007-06-14 19:39:12 +00:00
ast_debug ( 1 , " ast call on peer returned %d \n " , res ) ;
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " Couldn't call %s \n " , numsubst ) ;
2008-11-20 17:39:06 +00:00
if ( tc - > hangupcause ) {
chan - > hangupcause = tc - > hangupcause ;
}
2009-04-03 22:41:46 +00:00
ast_channel_unlock ( chan ) ;
2006-12-19 09:58:40 +00:00
ast_hangup ( tc ) ;
tc = NULL ;
2007-06-06 21:20:11 +00:00
ast_free ( tmp ) ;
1999-12-04 21:35:07 +00:00
continue ;
2005-01-16 07:58:51 +00:00
} else {
2009-04-09 17:39:10 +00:00
const char * tmpexten = ast_strdupa ( S_OR ( chan - > macroexten , chan - > exten ) ) ;
2007-12-06 15:04:34 +00:00
senddialevent ( chan , tc , numsubst ) ;
2007-07-26 15:49:18 +00:00
ast_verb ( 3 , " Called %s \n " , numsubst ) ;
2009-04-09 20:40:34 +00:00
ast_channel_unlock ( chan ) ;
2009-04-03 22:41:46 +00:00
if ( ! ast_test_flag64 ( peerflags , OPT_ORIGINAL_CLID ) ) {
2009-04-09 17:39:10 +00:00
ast_set_callerid ( tc , tmpexten , get_cid_name ( cidname , sizeof ( cidname ) , chan ) , NULL ) ;
2009-04-03 22:41:46 +00:00
}
2005-01-16 07:58:51 +00:00
}
2008-02-09 11:27:10 +00:00
/* Put them in the list of outgoing thingies... We're ready now.
1999-12-04 21:35:07 +00:00
XXX If we ' re forcibly removed , these outgoing calls won ' t get
hung up XXX */
2008-02-09 11:27:10 +00:00
ast_set_flag64 ( tmp , DIAL_STILLGOING ) ;
2006-12-19 09:58:40 +00:00
tmp - > chan = tc ;
1999-12-04 21:35:07 +00:00
tmp - > next = outgoing ;
outgoing = tmp ;
2002-09-02 15:20:28 +00:00
/* If this line is up, don't try anybody else */
if ( outgoing - > chan - > _state = = AST_STATE_UP )
break ;
2006-04-19 14:02:49 +00:00
}
2001-03-30 18:47:35 +00:00
2006-04-19 14:02:49 +00:00
if ( ast_strlen_zero ( args . timeout ) ) {
to = - 1 ;
} else {
2005-11-02 21:46:52 +00:00
to = atoi ( args . timeout ) ;
2004-04-02 07:47:23 +00:00
if ( to > 0 )
to * = 1000 ;
2008-10-14 23:57:46 +00:00
else {
ast_log ( LOG_WARNING , " Invalid timeout specified: '%s'. Setting timeout to infinite \n " , args . timeout ) ;
to = - 1 ;
}
2006-04-19 14:02:49 +00:00
}
2004-06-21 18:28:35 +00:00
2006-04-19 14:02:49 +00:00
if ( ! outgoing ) {
2006-11-04 11:00:49 +00:00
strcpy ( pa . status , " CHANUNAVAIL " ) ;
2007-12-12 20:05:13 +00:00
if ( fulldial = = num_dialed ) {
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
res = - 1 ;
goto out ;
}
2006-04-19 14:02:49 +00:00
} else {
2004-06-23 03:16:58 +00:00
/* Our status will at least be NOANSWER */
2006-11-04 11:00:49 +00:00
strcpy ( pa . status , " NOANSWER " ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( outgoing , OPT_MUSICBACK ) ) {
2006-07-19 20:44:39 +00:00
moh = 1 ;
2007-02-09 19:39:26 +00:00
if ( ! ast_strlen_zero ( opt_args [ OPT_ARG_MUSICBACK ] ) ) {
char * original_moh = ast_strdupa ( chan - > musicclass ) ;
ast_string_field_set ( chan , musicclass , opt_args [ OPT_ARG_MUSICBACK ] ) ;
ast_moh_start ( chan , opt_args [ OPT_ARG_MUSICBACK ] , NULL ) ;
ast_string_field_set ( chan , musicclass , original_moh ) ;
} else {
ast_moh_start ( chan , NULL , NULL ) ;
}
2006-12-01 23:39:59 +00:00
ast_indicate ( chan , AST_CONTROL_PROGRESS ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
} else if ( ast_test_flag64 ( outgoing , OPT_RINGBACK ) ) {
2004-06-22 13:53:45 +00:00
ast_indicate ( chan , AST_CONTROL_RINGING ) ;
sentringing + + ;
}
2006-04-19 14:02:49 +00:00
}
2004-06-21 18:28:35 +00:00
2009-03-17 17:17:51 +00:00
peer = wait_for_answer ( chan , outgoing , & to , peerflags , & pa , & num , & result , dtmf_progress ) ;
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
2008-04-14 16:25:09 +00:00
/* The ast_channel_datastore_remove() function could fail here if the
* datastore was moved to another channel during a masquerade . If this is
* the case , don ' t free the datastore here because later , when the channel
* to which the datastore was moved hangs up , it will attempt to free this
* datastore again , causing a crash
*/
if ( ! ast_channel_datastore_remove ( chan , datastore ) )
2008-08-05 16:56:11 +00:00
ast_datastore_free ( datastore ) ;
1999-12-04 21:35:07 +00:00
if ( ! peer ) {
2005-01-18 03:12:53 +00:00
if ( result ) {
res = result ;
2006-04-19 14:02:49 +00:00
} else if ( to ) { /* Musta gotten hung up */
1999-12-04 21:35:07 +00:00
res = - 1 ;
2006-04-19 14:02:49 +00:00
} else { /* Nobody answered, next please? */
2005-09-07 19:13:00 +00:00
res = 0 ;
2006-04-19 14:02:49 +00:00
}
2008-04-28 16:37:45 +00:00
/* SIP, in particular, sends back this error code to indicate an
* overlap dialled number needs more digits . */
if ( chan - > hangupcause = = AST_CAUSE_INVALID_NUMBER_FORMAT ) {
res = AST_PBX_INCOMPLETE ;
}
2006-04-19 16:54:04 +00:00
/* almost done, although the 'else' block is 400 lines */
2006-04-19 14:02:49 +00:00
} else {
2006-04-19 16:36:15 +00:00
const char * number ;
2006-11-04 11:00:49 +00:00
strcpy ( pa . status , " ANSWER " ) ;
2008-10-31 18:55:33 +00:00
pbx_builtin_setvar_helper ( chan , " DIALSTATUS " , pa . status ) ;
1999-12-04 21:35:07 +00:00
/* Ah ha! Someone answered within the desired timeframe. Of course after this
2008-02-09 11:27:10 +00:00
we will always return with - 1 so that it is hung up properly after the
1999-12-04 21:35:07 +00:00
conversation . */
2007-07-09 08:27:37 +00:00
hanguptree ( outgoing , peer , 1 ) ;
1999-12-04 21:35:07 +00:00
outgoing = NULL ;
2001-12-20 15:21:47 +00:00
/* If appropriate, log that we have a destination channel */
if ( chan - > cdr )
ast_cdr_setdestchan ( chan - > cdr , peer - > name ) ;
2003-03-12 06:00:18 +00:00
if ( peer - > name )
pbx_builtin_setvar_helper ( chan , " DIALEDPEERNAME " , peer - > name ) ;
2008-06-18 13:09:02 +00:00
ast_channel_lock ( peer ) ;
number = pbx_builtin_getvar_helper ( peer , " DIALEDPEERNUMBER " ) ;
2005-01-27 16:33:12 +00:00
if ( ! number )
number = numsubst ;
pbx_builtin_setvar_helper ( chan , " DIALEDPEERNUMBER " , number ) ;
2008-06-18 13:09:02 +00:00
ast_channel_unlock ( peer ) ;
2008-02-09 11:27:10 +00:00
if ( ! ast_strlen_zero ( args . url ) & & ast_channel_supports_html ( peer ) ) {
ast_debug ( 1 , " app_dial: sendurl=%s. \n " , args . url ) ;
ast_channel_sendurl ( peer , args . url ) ;
}
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ( ast_test_flag64 ( & opts , OPT_PRIVACY ) | | ast_test_flag64 ( & opts , OPT_SCREENING ) ) & & pa . privdb_val = = AST_PRIVACY_UNKNOWN ) {
2006-11-04 11:00:49 +00:00
if ( do_privacy ( chan , peer , & opts , opt_args , & pa ) ) {
2006-04-19 18:00:32 +00:00
res = 0 ;
goto out ;
}
2005-07-12 03:23:31 +00:00
}
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ! ast_test_flag64 ( & opts , OPT_ANNOUNCE ) | | ast_strlen_zero ( opt_args [ OPT_ARG_ANNOUNCE ] ) ) {
2006-04-19 14:02:49 +00:00
res = 0 ;
} else {
2006-04-19 16:54:04 +00:00
int digit = 0 ;
2004-12-19 21:13:41 +00:00
/* Start autoservice on the other chan */
2004-05-07 20:39:14 +00:00
res = ast_autoservice_start ( chan ) ;
2004-12-19 21:13:41 +00:00
/* Now Stream the File */
2004-05-07 20:39:14 +00:00
if ( ! res )
2005-11-02 21:46:52 +00:00
res = ast_streamfile ( peer , opt_args [ OPT_ARG_ANNOUNCE ] , peer - > language ) ;
2004-05-20 00:29:09 +00:00
if ( ! res ) {
2008-02-09 11:27:10 +00:00
digit = ast_waitstream ( peer , AST_DIGIT_ANY ) ;
2004-05-20 00:29:09 +00:00
}
2004-12-19 21:13:41 +00:00
/* Ok, done. stop autoservice */
2004-05-07 20:39:14 +00:00
res = ast_autoservice_stop ( chan ) ;
2004-05-20 00:29:09 +00:00
if ( digit > 0 & & ! res )
2008-02-09 11:27:10 +00:00
res = ast_senddigit ( chan , digit , 0 ) ;
2004-05-20 00:29:09 +00:00
else
res = digit ;
2006-04-19 14:02:49 +00:00
}
2005-03-17 22:39:04 +00:00
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( chan & & peer & & ast_test_flag64 ( & opts , OPT_GOTO ) & & ! ast_strlen_zero ( opt_args [ OPT_ARG_GOTO ] ) ) {
2006-04-19 14:02:49 +00:00
replace_macro_delimiter ( opt_args [ OPT_ARG_GOTO ] ) ;
2005-11-02 21:46:52 +00:00
ast_parseable_goto ( chan , opt_args [ OPT_ARG_GOTO ] ) ;
2007-12-05 22:55:49 +00:00
/* peer goes to the same context and extension as chan, so just copy info from chan*/
ast_copy_string ( peer - > context , chan - > context , sizeof ( peer - > context ) ) ;
ast_copy_string ( peer - > exten , chan - > exten , sizeof ( peer - > exten ) ) ;
peer - > priority = chan - > priority + 2 ;
2005-03-17 22:39:04 +00:00
ast_pbx_start ( peer ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
hanguptree ( outgoing , NULL , ast_test_flag64 ( & opts , OPT_CANCEL_ELSEWHERE ) ? 1 : 0 ) ;
2007-02-15 16:24:13 +00:00
if ( continue_exec )
* continue_exec = 1 ;
res = 0 ;
2006-04-19 14:02:49 +00:00
goto done ;
2005-03-17 22:39:04 +00:00
}
2003-07-02 14:06:12 +00:00
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( & opts , OPT_CALLEE_MACRO ) & & ! ast_strlen_zero ( opt_args [ OPT_ARG_CALLEE_MACRO ] ) ) {
2006-04-19 16:54:04 +00:00
struct ast_app * theapp ;
2006-04-19 18:07:19 +00:00
const char * macro_result ;
2006-04-19 16:54:04 +00:00
2004-09-10 02:31:30 +00:00
res = ast_autoservice_start ( chan ) ;
if ( res ) {
ast_log ( LOG_ERROR , " Unable to start autoservice on calling channel \n " ) ;
res = - 1 ;
}
2006-04-19 16:54:04 +00:00
theapp = pbx_findapp ( " Macro " ) ;
2004-09-10 02:31:30 +00:00
2008-02-09 11:27:10 +00:00
if ( theapp & & ! res ) { /* XXX why check res here ? */
2009-01-09 00:13:12 +00:00
/* Set peer->exten and peer->context so that MACRO_EXTEN and MACRO_CONTEXT get set */
ast_copy_string ( peer - > context , chan - > context , sizeof ( peer - > context ) ) ;
ast_copy_string ( peer - > exten , chan - > exten , sizeof ( peer - > exten ) ) ;
2006-04-19 14:02:49 +00:00
replace_macro_delimiter ( opt_args [ OPT_ARG_CALLEE_MACRO ] ) ;
2006-04-19 16:54:04 +00:00
res = pbx_exec ( peer , theapp , opt_args [ OPT_ARG_CALLEE_MACRO ] ) ;
2007-06-14 19:39:12 +00:00
ast_debug ( 1 , " Macro exited with status %d \n " , res ) ;
2004-09-10 02:31:30 +00:00
res = 0 ;
} else {
ast_log ( LOG_ERROR , " Could not find application Macro \n " ) ;
res = - 1 ;
}
if ( ast_autoservice_stop ( chan ) < 0 ) {
ast_log ( LOG_ERROR , " Could not stop autoservice on calling channel \n " ) ;
res = - 1 ;
}
2004-11-22 22:11:10 +00:00
2008-06-18 13:09:02 +00:00
ast_channel_lock ( peer ) ;
2006-04-19 18:07:19 +00:00
if ( ! res & & ( macro_result = pbx_builtin_getvar_helper ( peer , " MACRO_RESULT " ) ) ) {
2006-11-04 01:16:20 +00:00
char * macro_transfer_dest ;
2005-01-05 23:05:49 +00:00
2006-11-04 01:16:20 +00:00
if ( ! strcasecmp ( macro_result , " BUSY " ) ) {
2006-11-04 11:00:49 +00:00
ast_copy_string ( pa . status , macro_result , sizeof ( pa . status ) ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
ast_set_flag64 ( peerflags , OPT_GO_ON ) ;
2006-11-04 01:16:20 +00:00
res = - 1 ;
} else if ( ! strcasecmp ( macro_result , " CONGESTION " ) | | ! strcasecmp ( macro_result , " CHANUNAVAIL " ) ) {
2006-11-04 11:00:49 +00:00
ast_copy_string ( pa . status , macro_result , sizeof ( pa . status ) ) ;
2008-02-09 11:27:10 +00:00
ast_set_flag64 ( peerflags , OPT_GO_ON ) ;
2006-11-04 01:16:20 +00:00
res = - 1 ;
} else if ( ! strcasecmp ( macro_result , " CONTINUE " ) ) {
2008-02-09 11:27:10 +00:00
/* hangup peer and keep chan alive assuming the macro has changed
the context / exten / priority or perhaps
2006-11-04 01:16:20 +00:00
the next priority in the current exten is desired .
*/
2008-02-09 11:27:10 +00:00
ast_set_flag64 ( peerflags , OPT_GO_ON ) ;
2006-11-04 01:16:20 +00:00
res = - 1 ;
} else if ( ! strcasecmp ( macro_result , " ABORT " ) ) {
/* Hangup both ends unless the caller has the g flag */
res = - 1 ;
} else if ( ! strncasecmp ( macro_result , " GOTO: " , 5 ) & & ( macro_transfer_dest = ast_strdupa ( macro_result + 5 ) ) ) {
res = - 1 ;
/* perform a transfer to a new extension */
if ( strchr ( macro_transfer_dest , ' ^ ' ) ) { /* context^exten^priority*/
replace_macro_delimiter ( macro_transfer_dest ) ;
if ( ! ast_parseable_goto ( chan , macro_transfer_dest ) )
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
ast_set_flag64 ( peerflags , OPT_GO_ON ) ;
2004-11-22 22:11:10 +00:00
}
2006-11-04 01:16:20 +00:00
}
2004-11-22 22:11:10 +00:00
}
2008-06-18 13:09:02 +00:00
ast_channel_unlock ( peer ) ;
2004-09-10 02:31:30 +00:00
}
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( & opts , OPT_CALLEE_GOSUB ) & & ! ast_strlen_zero ( opt_args [ OPT_ARG_CALLEE_GOSUB ] ) ) {
2007-06-19 23:36:34 +00:00
struct ast_app * theapp ;
const char * gosub_result ;
2007-06-20 17:35:08 +00:00
char * gosub_args , * gosub_argstart ;
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
int res9 = - 1 ;
2007-06-19 23:36:34 +00:00
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
res9 = ast_autoservice_start ( chan ) ;
if ( res9 ) {
2007-06-19 23:36:34 +00:00
ast_log ( LOG_ERROR , " Unable to start autoservice on calling channel \n " ) ;
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
res9 = - 1 ;
2007-06-19 23:36:34 +00:00
}
theapp = pbx_findapp ( " Gosub " ) ;
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
if ( theapp & & ! res9 ) {
2007-06-19 23:36:34 +00:00
replace_macro_delimiter ( opt_args [ OPT_ARG_CALLEE_GOSUB ] ) ;
2007-06-20 17:35:08 +00:00
/* Set where we came from */
ast_copy_string ( peer - > context , " app_dial_gosub_virtual_context " , sizeof ( peer - > context ) ) ;
ast_copy_string ( peer - > exten , " s " , sizeof ( peer - > exten ) ) ;
peer - > priority = 0 ;
2008-05-30 16:10:46 +00:00
gosub_argstart = strchr ( opt_args [ OPT_ARG_CALLEE_GOSUB ] , ' , ' ) ;
2007-06-20 17:35:08 +00:00
if ( gosub_argstart ) {
* gosub_argstart = 0 ;
2008-11-02 18:52:13 +00:00
if ( asprintf ( & gosub_args , " %s,s,1(%s) " , opt_args [ OPT_ARG_CALLEE_GOSUB ] , gosub_argstart + 1 ) < 0 ) {
ast_log ( LOG_WARNING , " asprintf() failed: %s \n " , strerror ( errno ) ) ;
gosub_args = NULL ;
}
2008-05-30 16:10:46 +00:00
* gosub_argstart = ' , ' ;
2007-06-20 17:35:08 +00:00
} else {
2008-11-02 18:52:13 +00:00
if ( asprintf ( & gosub_args , " %s,s,1 " , opt_args [ OPT_ARG_CALLEE_GOSUB ] ) < 0 ) {
ast_log ( LOG_WARNING , " asprintf() failed: %s \n " , strerror ( errno ) ) ;
gosub_args = NULL ;
}
2007-06-20 17:35:08 +00:00
}
2007-06-20 21:38:49 +00:00
2007-06-20 17:35:08 +00:00
if ( gosub_args ) {
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
res9 = pbx_exec ( peer , theapp , gosub_args ) ;
2008-12-03 18:37:46 +00:00
if ( ! res9 ) {
2008-12-18 19:33:42 +00:00
struct ast_pbx_args args ;
/* A struct initializer fails to compile for this case ... */
memset ( & args , 0 , sizeof ( args ) ) ;
args . no_hangup_chan = 1 ;
ast_pbx_run_args ( peer , & args ) ;
2008-12-03 18:37:46 +00:00
}
2007-12-12 20:05:13 +00:00
ast_free ( gosub_args ) ;
2008-12-29 18:04:52 +00:00
ast_debug ( 1 , " Gosub exited with status %d \n " , res9 ) ;
2008-12-03 18:37:46 +00:00
} else {
2007-06-20 17:35:08 +00:00
ast_log ( LOG_ERROR , " Could not Allocate string for Gosub arguments -- Gosub Call Aborted! \n " ) ;
2008-12-03 18:37:46 +00:00
}
2008-02-09 11:27:10 +00:00
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
} else if ( ! res9 ) {
2007-06-19 23:36:34 +00:00
ast_log ( LOG_ERROR , " Could not find application Gosub \n " ) ;
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
res9 = - 1 ;
2007-06-19 23:36:34 +00:00
}
if ( ast_autoservice_stop ( chan ) < 0 ) {
ast_log ( LOG_ERROR , " Could not stop autoservice on calling channel \n " ) ;
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
res9 = - 1 ;
2007-06-19 23:36:34 +00:00
}
2008-06-18 13:09:02 +00:00
ast_channel_lock ( peer ) ;
2007-06-19 23:36:34 +00:00
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
if ( ! res9 & & ( gosub_result = pbx_builtin_getvar_helper ( peer , " GOSUB_RESULT " ) ) ) {
2007-06-19 23:36:34 +00:00
char * gosub_transfer_dest ;
if ( ! strcasecmp ( gosub_result , " BUSY " ) ) {
ast_copy_string ( pa . status , gosub_result , sizeof ( pa . status ) ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
ast_set_flag64 ( peerflags , OPT_GO_ON ) ;
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
res9 = - 1 ;
2007-06-19 23:36:34 +00:00
} else if ( ! strcasecmp ( gosub_result , " CONGESTION " ) | | ! strcasecmp ( gosub_result , " CHANUNAVAIL " ) ) {
ast_copy_string ( pa . status , gosub_result , sizeof ( pa . status ) ) ;
2008-02-09 11:27:10 +00:00
ast_set_flag64 ( peerflags , OPT_GO_ON ) ;
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
res9 = - 1 ;
2007-06-19 23:36:34 +00:00
} else if ( ! strcasecmp ( gosub_result , " CONTINUE " ) ) {
2008-02-09 11:27:10 +00:00
/* hangup peer and keep chan alive assuming the macro has changed
the context / exten / priority or perhaps
2007-06-19 23:36:34 +00:00
the next priority in the current exten is desired .
*/
2008-02-09 11:27:10 +00:00
ast_set_flag64 ( peerflags , OPT_GO_ON ) ;
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
res9 = - 1 ;
2007-06-19 23:36:34 +00:00
} else if ( ! strcasecmp ( gosub_result , " ABORT " ) ) {
/* Hangup both ends unless the caller has the g flag */
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
res9 = - 1 ;
2007-06-19 23:36:34 +00:00
} else if ( ! strncasecmp ( gosub_result , " GOTO: " , 5 ) & & ( gosub_transfer_dest = ast_strdupa ( gosub_result + 5 ) ) ) {
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
res9 = - 1 ;
2007-06-19 23:36:34 +00:00
/* perform a transfer to a new extension */
if ( strchr ( gosub_transfer_dest , ' ^ ' ) ) { /* context^exten^priority*/
replace_macro_delimiter ( gosub_transfer_dest ) ;
if ( ! ast_parseable_goto ( chan , gosub_transfer_dest ) )
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
ast_set_flag64 ( peerflags , OPT_GO_ON ) ;
2007-06-19 23:36:34 +00:00
}
}
}
2008-06-18 13:09:02 +00:00
ast_channel_unlock ( peer ) ;
2007-06-19 23:36:34 +00:00
}
2004-05-07 21:14:55 +00:00
if ( ! res ) {
2008-11-12 21:34:51 +00:00
if ( ! ast_tvzero ( calldurationlimit ) ) {
struct timeval whentohangup = calldurationlimit ;
2008-05-01 23:06:23 +00:00
peer - > whentohangup = ast_tvadd ( ast_tvnow ( ) , whentohangup ) ;
2004-05-07 21:14:55 +00:00
}
2008-02-09 11:27:10 +00:00
if ( ! ast_strlen_zero ( dtmfcalled ) ) {
2008-02-05 23:00:15 +00:00
ast_verb ( 3 , " Sending DTMF '%s' to the called party. \n " , dtmfcalled ) ;
2007-12-12 20:05:13 +00:00
res = ast_dtmf_stream ( peer , chan , dtmfcalled , 250 , 0 ) ;
2005-04-11 02:46:25 +00:00
}
2005-10-26 19:48:14 +00:00
if ( ! ast_strlen_zero ( dtmfcalling ) ) {
2008-02-05 23:00:15 +00:00
ast_verb ( 3 , " Sending DTMF '%s' to the calling party. \n " , dtmfcalling ) ;
2007-12-12 20:05:13 +00:00
res = ast_dtmf_stream ( chan , peer , dtmfcalling , 250 , 0 ) ;
2005-04-11 02:46:25 +00:00
}
2004-05-07 20:39:14 +00:00
}
2008-10-31 18:55:33 +00:00
2008-02-09 11:27:10 +00:00
if ( res ) { /* some error */
2006-11-04 01:16:20 +00:00
res = - 1 ;
} else {
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( peerflags , OPT_CALLEE_TRANSFER ) )
2005-01-10 14:46:59 +00:00
ast_set_flag ( & ( config . features_callee ) , AST_FEATURE_REDIRECT ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( peerflags , OPT_CALLER_TRANSFER ) )
2005-01-10 14:46:59 +00:00
ast_set_flag ( & ( config . features_caller ) , AST_FEATURE_REDIRECT ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( peerflags , OPT_CALLEE_HANGUP ) )
2005-01-10 14:46:59 +00:00
ast_set_flag ( & ( config . features_callee ) , AST_FEATURE_DISCONNECT ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( peerflags , OPT_CALLER_HANGUP ) )
2005-01-10 14:46:59 +00:00
ast_set_flag ( & ( config . features_caller ) , AST_FEATURE_DISCONNECT ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( peerflags , OPT_CALLEE_MONITOR ) )
2005-01-10 14:46:59 +00:00
ast_set_flag ( & ( config . features_callee ) , AST_FEATURE_AUTOMON ) ;
2008-02-09 11:27:10 +00:00
if ( ast_test_flag64 ( peerflags , OPT_CALLER_MONITOR ) )
2005-01-10 14:46:59 +00:00
ast_set_flag ( & ( config . features_caller ) , AST_FEATURE_AUTOMON ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( peerflags , OPT_CALLEE_PARK ) )
2006-05-22 16:43:43 +00:00
ast_set_flag ( & ( config . features_callee ) , AST_FEATURE_PARKCALL ) ;
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( peerflags , OPT_CALLER_PARK ) )
2006-05-22 16:43:43 +00:00
ast_set_flag ( & ( config . features_caller ) , AST_FEATURE_PARKCALL ) ;
2007-11-30 21:19:57 +00:00
if ( ast_test_flag64 ( peerflags , OPT_CALLEE_MIXMONITOR ) )
ast_set_flag ( & ( config . features_callee ) , AST_FEATURE_AUTOMIXMON ) ;
if ( ast_test_flag64 ( peerflags , OPT_CALLER_MIXMONITOR ) )
ast_set_flag ( & ( config . features_caller ) , AST_FEATURE_AUTOMIXMON ) ;
Merged revisions 142675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines
Tested by: sergee, murf, chris-mac, andrew, KNK
This is a "second attempt" to restore the previous "endbeforeh" behavior
in 1.4 and up. In order to capture information concerning all the
legs of transfers in all their infinite combinations, I was forced
to this particular solution by a chain of logical necessities, the
first being that I was not allowed to rewrite the CDR mechanism from
the ground up!
This change basically leaves the original machinery alone, which allows
IVR and local channel type situations to generate CDR's as normal, but
a channel flag can be set to suppress the normal running of the h exten.
That flag would be set by the code that runs the h exten from the
ast_bridge_call routine, to prevent the h exten from being run twice.
Also, a flag in the ast_bridge_config struct passed into ast_bridge_call
can be used to suppress the running of the h exten in that routine. This
would happen, for instance, if you use the 'g' option in the Dial app.
Running this routine 'early' allows not only the CDR() func to be used
in the h extension for reading CDR variables, but also allows them to
be modified before the CDR is posted to the backends.
While I dearly hope that this patch overcomes all problems, and
introduces no new problems, reality suggests that surely someone
will have problems. In this case, please re-open 13251 (or 13289),
and we'll see if we can't fix any remaining issues.
** trunk note: some code to suppress the h exten being run
from app_queue was added; for the 'continue' option available
only in trunk/1.6.x.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12 04:50:48 +00:00
if ( ast_test_flag64 ( peerflags , OPT_GO_ON ) )
ast_set_flag ( & ( config . features_caller ) , AST_FEATURE_NO_H_EXTEN ) ;
2005-01-18 03:12:53 +00:00
2008-10-31 18:55:33 +00:00
config . end_bridge_callback = end_bridge_callback ;
2008-11-09 01:27:00 +00:00
config . end_bridge_callback_data = chan ;
2008-11-18 18:31:08 +00:00
config . end_bridge_callback_data_fixup = end_bridge_callback_data_fixup ;
2004-06-21 18:28:35 +00:00
if ( moh ) {
moh = 0 ;
ast_moh_stop ( chan ) ;
} else if ( sentringing ) {
sentringing = 0 ;
ast_indicate ( chan , - 1 ) ;
}
2004-07-07 16:02:13 +00:00
/* Be sure no generators are left on it */
ast_deactivate_generator ( chan ) ;
/* Make sure channels are compatible */
res = ast_channel_make_compatible ( chan , peer ) ;
if ( res < 0 ) {
ast_log ( LOG_WARNING , " Had to drop call because I couldn't make %s compatible with %s \n " , chan - > name , peer - > name ) ;
ast_hangup ( peer ) ;
2006-04-19 16:54:04 +00:00
res = - 1 ;
goto done ;
2004-07-07 16:02:13 +00:00
}
2008-10-07 21:34:44 +00:00
if ( opermode ) {
2006-04-22 11:30:06 +00:00
struct oprmode oprmode ;
oprmode . peer = peer ;
oprmode . mode = opermode ;
2007-12-14 14:48:38 +00:00
ast_channel_setoption ( chan , AST_OPTION_OPRMODE , & oprmode , sizeof ( oprmode ) , 0 ) ;
2006-04-22 11:30:06 +00:00
}
2007-12-12 20:05:13 +00:00
res = ast_bridge_call ( chan , peer , & config ) ;
2006-04-19 16:19:52 +00:00
}
2006-11-04 01:16:20 +00:00
Merged revisions 166093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
In order to merge this 1.4 patch into trunk,
I had to resolve some conflicts and wait for
Russell to make some changes to res_agi.
I re-ran all the tests; 39 calls in all, and
made fairly careful notes and comparisons: I
don't want this to blow up some aspect of
asterisk; I completely removed the KEEPALIVE
from the pbx.h decls. The first 3 scenarios
involving feature park; feature xfer to 700;
hookflash park to Park() app call all behave
the same, don't appear to leave hung channels,
and no crashes.
........
r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
This merges the masqpark branch into 1.4
These changes eliminate the need for (and use of)
the KEEPALIVE return code in res_features.c;
There are other places that use this result code
for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk.
The reason these changes are being made in 1.4, is
that parking ends a channel's life, in some situations,
and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing
the channel pointer, which could lead to memory corruption
and crashes.
Calling the masq_park function eliminates this danger
in higher levels.
A series of previous commits have replaced some parking calls
with masq_park, but this patch puts them ALL to rest,
(except one, purposely left alone because a masquerade
is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes.
While bug 13820 inspired this work, this patch does
not solve all the problems mentioned there.
I have tested this patch (again) to make sure I have
not introduced regressions.
Crashes that occurred when a parked party hung up
while the parking party was listening to the numbers
of the parking stall being assigned, is eliminated.
These are the cases where parking code may be activated:
1. Feature one touch (eg. *3)
2. Feature blind xfer to parking lot (eg ##700)
3. Run Park() app from dialplan (eg sip xfer to 700)
(eg. dahdi hookflash xfer to 700)
4. Run Park via manager.
The interesting testing cases for parking are:
I. A calls B, A parks B
a. B hangs up while A is getting the numbers announced.
b. B hangs up after A gets the announcement, but
before the parking time expires
c. B waits, time expires, A is redialed,
A answers, B and A are connected, after
which, B hangs up.
d. C picks up B while still in parking lot.
II. A calls B, B parks A
a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but
before the parking time expires
c. A waits, time expires, B is redialed,
B answers, A and B are connected, after
which, A hangs up.
d. C picks up A while still in parking lot.
Testing this throroughly involves acting all the permutations
of I and II, in situations 1,2,3, and 4.
Since I added a few more changes (ALL references to KEEPALIVE in the bridge
code eliimated (I missed one earlier), I retested
most of the above cases, and no crashes.
H-extension weirdness.
Current h-extension execution is not completely
correct for several of the cases.
For the case where A calls B, and A parks B, the
'h' exten is run on A's channel as soon as the park
is accomplished. This is expected behavior.
But when A calls B, and B parks A, this will be
current behavior:
After B parks A, B is hung up by the system, and
the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's...
Thus, if A is DAHDI/1, and B is DAHDI/2,
the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the
start/answer/end info will be those
relating to Channel A.
And, in the case where A is reconnected to
B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten
will be run at all.
In the case where C picks up A from the
parking lot, when either A or C hang up,
the h-exten will be run for the C channel.
CDR's are a separate issue, and not addressed
here.
As to WHY this strange behavior occurs,
the answer lies in the procedure followed
to accomplish handing over the channel
to the parking manager thread. This procedure
is called masquerading. In the process,
a duplicate copy of the channel is created,
and most of the active data is given to the
new copy. The original channel gets its name
changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original
thread (preserving its role as a call
originator, if it had this role to begin
with), while the new channel is without
this info and becomes a call target (a
"peer").
In this case, the parking lot manager
thread is handed the new (masqueraded)
channel. It will not run an h-exten
on the channel if it hangs up while
in the parking lot. The h exten will
be run on the original channel instead,
in the original thread, after the bridge
completes.
See bug 13820 for our intentions as
to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23 18:13:49 +00:00
strcpy ( peer - > context , chan - > context ) ;
2008-02-09 11:27:10 +00:00
Merged revisions 166093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
In order to merge this 1.4 patch into trunk,
I had to resolve some conflicts and wait for
Russell to make some changes to res_agi.
I re-ran all the tests; 39 calls in all, and
made fairly careful notes and comparisons: I
don't want this to blow up some aspect of
asterisk; I completely removed the KEEPALIVE
from the pbx.h decls. The first 3 scenarios
involving feature park; feature xfer to 700;
hookflash park to Park() app call all behave
the same, don't appear to leave hung channels,
and no crashes.
........
r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
This merges the masqpark branch into 1.4
These changes eliminate the need for (and use of)
the KEEPALIVE return code in res_features.c;
There are other places that use this result code
for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk.
The reason these changes are being made in 1.4, is
that parking ends a channel's life, in some situations,
and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing
the channel pointer, which could lead to memory corruption
and crashes.
Calling the masq_park function eliminates this danger
in higher levels.
A series of previous commits have replaced some parking calls
with masq_park, but this patch puts them ALL to rest,
(except one, purposely left alone because a masquerade
is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes.
While bug 13820 inspired this work, this patch does
not solve all the problems mentioned there.
I have tested this patch (again) to make sure I have
not introduced regressions.
Crashes that occurred when a parked party hung up
while the parking party was listening to the numbers
of the parking stall being assigned, is eliminated.
These are the cases where parking code may be activated:
1. Feature one touch (eg. *3)
2. Feature blind xfer to parking lot (eg ##700)
3. Run Park() app from dialplan (eg sip xfer to 700)
(eg. dahdi hookflash xfer to 700)
4. Run Park via manager.
The interesting testing cases for parking are:
I. A calls B, A parks B
a. B hangs up while A is getting the numbers announced.
b. B hangs up after A gets the announcement, but
before the parking time expires
c. B waits, time expires, A is redialed,
A answers, B and A are connected, after
which, B hangs up.
d. C picks up B while still in parking lot.
II. A calls B, B parks A
a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but
before the parking time expires
c. A waits, time expires, B is redialed,
B answers, A and B are connected, after
which, A hangs up.
d. C picks up A while still in parking lot.
Testing this throroughly involves acting all the permutations
of I and II, in situations 1,2,3, and 4.
Since I added a few more changes (ALL references to KEEPALIVE in the bridge
code eliimated (I missed one earlier), I retested
most of the above cases, and no crashes.
H-extension weirdness.
Current h-extension execution is not completely
correct for several of the cases.
For the case where A calls B, and A parks B, the
'h' exten is run on A's channel as soon as the park
is accomplished. This is expected behavior.
But when A calls B, and B parks A, this will be
current behavior:
After B parks A, B is hung up by the system, and
the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's...
Thus, if A is DAHDI/1, and B is DAHDI/2,
the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the
start/answer/end info will be those
relating to Channel A.
And, in the case where A is reconnected to
B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten
will be run at all.
In the case where C picks up A from the
parking lot, when either A or C hang up,
the h-exten will be run for the C channel.
CDR's are a separate issue, and not addressed
here.
As to WHY this strange behavior occurs,
the answer lies in the procedure followed
to accomplish handing over the channel
to the parking manager thread. This procedure
is called masquerading. In the process,
a duplicate copy of the channel is created,
and most of the active data is given to the
new copy. The original channel gets its name
changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original
thread (preserving its role as a call
originator, if it had this role to begin
with), while the new channel is without
this info and becomes a call target (a
"peer").
In this case, the parking lot manager
thread is handed the new (masqueraded)
channel. It will not run an h-exten
on the channel if it hangs up while
in the parking lot. The h exten will
be run on the original channel instead,
in the original thread, after the bridge
completes.
See bug 13820 for our intentions as
to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23 18:13:49 +00:00
if ( ast_test_flag64 ( & opts , OPT_PEER_H ) & & ast_exists_extension ( peer , peer - > context , " h " , 1 , peer - > cid . cid_num ) ) {
2007-07-27 15:46:20 +00:00
int autoloopflag ;
This commits the performance mods that give the priority processing engine in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-01 22:26:51 +00:00
int found ;
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
int res9 ;
2007-07-17 19:40:29 +00:00
strcpy ( peer - > exten , " h " ) ;
peer - > priority = 1 ;
2008-02-09 11:27:10 +00:00
autoloopflag = ast_test_flag ( peer , AST_FLAG_IN_AUTOLOOP ) ; /* save value to restore at the end */
2007-07-27 15:46:20 +00:00
ast_set_flag ( peer , AST_FLAG_IN_AUTOLOOP ) ;
2008-02-09 11:27:10 +00:00
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
while ( ( res9 = ast_spawn_extension ( peer , peer - > context , peer - > exten , peer - > priority , peer - > cid . cid_num , & found , 1 ) ) = = 0 )
2007-07-17 19:40:29 +00:00
peer - > priority + + ;
2008-02-09 11:27:10 +00:00
Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
if ( found & & res9 ) {
This commits the performance mods that give the priority processing engine in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-01 22:26:51 +00:00
/* Something bad happened, or a hangup has been requested. */
ast_debug ( 1 , " Spawn extension (%s,%s,%d) exited non-zero on '%s' \n " , peer - > context , peer - > exten , peer - > priority , peer - > name ) ;
ast_verb ( 2 , " Spawn extension (%s, %s, %d) exited non-zero on '%s' \n " , peer - > context , peer - > exten , peer - > priority , peer - > name ) ;
}
2007-07-27 15:46:20 +00:00
ast_set2_flag ( peer , autoloopflag , AST_FLAG_IN_AUTOLOOP ) ; /* set it back the way it was */
2007-07-17 19:40:29 +00:00
}
2009-04-09 19:10:02 +00:00
if ( ! ast_check_hangup ( peer ) & & ast_test_flag64 ( & opts , OPT_CALLEE_GO_ON ) ) {
if ( ! ast_strlen_zero ( opt_args [ OPT_ARG_CALLEE_GO_ON ] ) ) {
replace_macro_delimiter ( opt_args [ OPT_ARG_CALLEE_GO_ON ] ) ;
ast_parseable_goto ( peer , opt_args [ OPT_ARG_CALLEE_GO_ON ] ) ;
} else { /* F() */
int res ;
res = ast_goto_if_exists ( peer , chan - > context , chan - > exten , ( chan - > priority ) + 1 ) ;
if ( res = = AST_PBX_GOTO_FAILED ) {
ast_hangup ( peer ) ;
goto out ;
}
}
Merged revisions 166093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
In order to merge this 1.4 patch into trunk,
I had to resolve some conflicts and wait for
Russell to make some changes to res_agi.
I re-ran all the tests; 39 calls in all, and
made fairly careful notes and comparisons: I
don't want this to blow up some aspect of
asterisk; I completely removed the KEEPALIVE
from the pbx.h decls. The first 3 scenarios
involving feature park; feature xfer to 700;
hookflash park to Park() app call all behave
the same, don't appear to leave hung channels,
and no crashes.
........
r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
This merges the masqpark branch into 1.4
These changes eliminate the need for (and use of)
the KEEPALIVE return code in res_features.c;
There are other places that use this result code
for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk.
The reason these changes are being made in 1.4, is
that parking ends a channel's life, in some situations,
and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing
the channel pointer, which could lead to memory corruption
and crashes.
Calling the masq_park function eliminates this danger
in higher levels.
A series of previous commits have replaced some parking calls
with masq_park, but this patch puts them ALL to rest,
(except one, purposely left alone because a masquerade
is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes.
While bug 13820 inspired this work, this patch does
not solve all the problems mentioned there.
I have tested this patch (again) to make sure I have
not introduced regressions.
Crashes that occurred when a parked party hung up
while the parking party was listening to the numbers
of the parking stall being assigned, is eliminated.
These are the cases where parking code may be activated:
1. Feature one touch (eg. *3)
2. Feature blind xfer to parking lot (eg ##700)
3. Run Park() app from dialplan (eg sip xfer to 700)
(eg. dahdi hookflash xfer to 700)
4. Run Park via manager.
The interesting testing cases for parking are:
I. A calls B, A parks B
a. B hangs up while A is getting the numbers announced.
b. B hangs up after A gets the announcement, but
before the parking time expires
c. B waits, time expires, A is redialed,
A answers, B and A are connected, after
which, B hangs up.
d. C picks up B while still in parking lot.
II. A calls B, B parks A
a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but
before the parking time expires
c. A waits, time expires, B is redialed,
B answers, A and B are connected, after
which, A hangs up.
d. C picks up A while still in parking lot.
Testing this throroughly involves acting all the permutations
of I and II, in situations 1,2,3, and 4.
Since I added a few more changes (ALL references to KEEPALIVE in the bridge
code eliimated (I missed one earlier), I retested
most of the above cases, and no crashes.
H-extension weirdness.
Current h-extension execution is not completely
correct for several of the cases.
For the case where A calls B, and A parks B, the
'h' exten is run on A's channel as soon as the park
is accomplished. This is expected behavior.
But when A calls B, and B parks A, this will be
current behavior:
After B parks A, B is hung up by the system, and
the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's...
Thus, if A is DAHDI/1, and B is DAHDI/2,
the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the
start/answer/end info will be those
relating to Channel A.
And, in the case where A is reconnected to
B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten
will be run at all.
In the case where C picks up A from the
parking lot, when either A or C hang up,
the h-exten will be run for the C channel.
CDR's are a separate issue, and not addressed
here.
As to WHY this strange behavior occurs,
the answer lies in the procedure followed
to accomplish handing over the channel
to the parking manager thread. This procedure
is called masquerading. In the process,
a duplicate copy of the channel is created,
and most of the active data is given to the
new copy. The original channel gets its name
changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original
thread (preserving its role as a call
originator, if it had this role to begin
with), while the new channel is without
this info and becomes a call target (a
"peer").
In this case, the parking lot manager
thread is handed the new (masqueraded)
channel. It will not run an h-exten
on the channel if it hangs up while
in the parking lot. The h exten will
be run on the original channel instead,
in the original thread, after the bridge
completes.
See bug 13820 for our intentions as
to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23 18:13:49 +00:00
ast_pbx_start ( peer ) ;
} else {
if ( ! ast_check_hangup ( chan ) )
chan - > hangupcause = peer - > hangupcause ;
ast_hangup ( peer ) ;
2004-05-22 23:17:33 +00:00
}
2008-02-09 11:27:10 +00:00
}
1999-12-04 21:35:07 +00:00
out :
Merged revisions 166093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
In order to merge this 1.4 patch into trunk,
I had to resolve some conflicts and wait for
Russell to make some changes to res_agi.
I re-ran all the tests; 39 calls in all, and
made fairly careful notes and comparisons: I
don't want this to blow up some aspect of
asterisk; I completely removed the KEEPALIVE
from the pbx.h decls. The first 3 scenarios
involving feature park; feature xfer to 700;
hookflash park to Park() app call all behave
the same, don't appear to leave hung channels,
and no crashes.
........
r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
This merges the masqpark branch into 1.4
These changes eliminate the need for (and use of)
the KEEPALIVE return code in res_features.c;
There are other places that use this result code
for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk.
The reason these changes are being made in 1.4, is
that parking ends a channel's life, in some situations,
and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing
the channel pointer, which could lead to memory corruption
and crashes.
Calling the masq_park function eliminates this danger
in higher levels.
A series of previous commits have replaced some parking calls
with masq_park, but this patch puts them ALL to rest,
(except one, purposely left alone because a masquerade
is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes.
While bug 13820 inspired this work, this patch does
not solve all the problems mentioned there.
I have tested this patch (again) to make sure I have
not introduced regressions.
Crashes that occurred when a parked party hung up
while the parking party was listening to the numbers
of the parking stall being assigned, is eliminated.
These are the cases where parking code may be activated:
1. Feature one touch (eg. *3)
2. Feature blind xfer to parking lot (eg ##700)
3. Run Park() app from dialplan (eg sip xfer to 700)
(eg. dahdi hookflash xfer to 700)
4. Run Park via manager.
The interesting testing cases for parking are:
I. A calls B, A parks B
a. B hangs up while A is getting the numbers announced.
b. B hangs up after A gets the announcement, but
before the parking time expires
c. B waits, time expires, A is redialed,
A answers, B and A are connected, after
which, B hangs up.
d. C picks up B while still in parking lot.
II. A calls B, B parks A
a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but
before the parking time expires
c. A waits, time expires, B is redialed,
B answers, A and B are connected, after
which, A hangs up.
d. C picks up A while still in parking lot.
Testing this throroughly involves acting all the permutations
of I and II, in situations 1,2,3, and 4.
Since I added a few more changes (ALL references to KEEPALIVE in the bridge
code eliimated (I missed one earlier), I retested
most of the above cases, and no crashes.
H-extension weirdness.
Current h-extension execution is not completely
correct for several of the cases.
For the case where A calls B, and A parks B, the
'h' exten is run on A's channel as soon as the park
is accomplished. This is expected behavior.
But when A calls B, and B parks A, this will be
current behavior:
After B parks A, B is hung up by the system, and
the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's...
Thus, if A is DAHDI/1, and B is DAHDI/2,
the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the
start/answer/end info will be those
relating to Channel A.
And, in the case where A is reconnected to
B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten
will be run at all.
In the case where C picks up A from the
parking lot, when either A or C hang up,
the h-exten will be run for the C channel.
CDR's are a separate issue, and not addressed
here.
As to WHY this strange behavior occurs,
the answer lies in the procedure followed
to accomplish handing over the channel
to the parking manager thread. This procedure
is called masquerading. In the process,
a duplicate copy of the channel is created,
and most of the active data is given to the
new copy. The original channel gets its name
changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original
thread (preserving its role as a call
originator, if it had this role to begin
with), while the new channel is without
this info and becomes a call target (a
"peer").
In this case, the parking lot manager
thread is handed the new (masqueraded)
channel. It will not run an h-exten
on the channel if it hangs up while
in the parking lot. The h exten will
be run on the original channel instead,
in the original thread, after the bridge
completes.
See bug 13820 for our intentions as
to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23 18:13:49 +00:00
if ( moh ) {
moh = 0 ;
ast_moh_stop ( chan ) ;
} else if ( sentringing ) {
sentringing = 0 ;
ast_indicate ( chan , - 1 ) ;
}
ast_channel_early_bridge ( chan , NULL ) ;
hanguptree ( outgoing , NULL , 0 ) ; /* In this case, there's no answer anywhere */
pbx_builtin_setvar_helper ( chan , " DIALSTATUS " , pa . status ) ;
senddialendevent ( chan , pa . status ) ;
ast_debug ( 1 , " Exiting with DIALSTATUS=%s. \n " , pa . status ) ;
if ( ( ast_test_flag64 ( peerflags , OPT_GO_ON ) ) & & ! ast_check_hangup ( chan ) & & ( res ! = AST_PBX_INCOMPLETE ) ) {
if ( ! ast_tvzero ( calldurationlimit ) )
memset ( & chan - > whentohangup , 0 , sizeof ( chan - > whentohangup ) ) ;
res = 0 ;
2007-06-07 14:23:21 +00:00
}
2006-04-19 14:02:49 +00:00
done :
2008-11-17 22:25:06 +00:00
if ( config . warning_sound ) {
ast_free ( ( char * ) config . warning_sound ) ;
}
if ( config . end_sound ) {
ast_free ( ( char * ) config . end_sound ) ;
}
if ( config . start_sound ) {
ast_free ( ( char * ) config . start_sound ) ;
}
1999-12-04 21:35:07 +00:00
return res ;
}
2005-01-18 03:12:53 +00:00
static int dial_exec ( struct ast_channel * chan , void * data )
{
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
struct ast_flags64 peerflags ;
2007-01-10 04:56:48 +00:00
2005-01-18 03:12:53 +00:00
memset ( & peerflags , 0 , sizeof ( peerflags ) ) ;
2007-01-10 04:56:48 +00:00
2007-02-15 16:24:13 +00:00
return dial_exec_full ( chan , data , & peerflags , NULL ) ;
2005-01-18 03:12:53 +00:00
}
static int retrydial_exec ( struct ast_channel * chan , void * data )
{
2007-07-23 19:51:41 +00:00
char * parse ;
2005-12-20 17:52:31 +00:00
const char * context = NULL ;
2008-08-10 14:45:25 +00:00
int sleepms = 0 , loops = 0 , res = - 1 ;
2007-07-23 19:51:41 +00:00
struct ast_flags64 peerflags = { 0 , } ;
AST_DECLARE_APP_ARGS ( args ,
AST_APP_ARG ( announce ) ;
AST_APP_ARG ( sleep ) ;
AST_APP_ARG ( retries ) ;
AST_APP_ARG ( dialdata ) ;
) ;
2005-10-26 19:48:14 +00:00
if ( ast_strlen_zero ( data ) ) {
2005-10-19 18:19:02 +00:00
ast_log ( LOG_WARNING , " RetryDial requires an argument! \n " ) ;
return - 1 ;
2008-02-09 11:27:10 +00:00
}
2005-01-18 03:12:53 +00:00
2007-07-23 19:51:41 +00:00
parse = ast_strdupa ( data ) ;
AST_STANDARD_APP_ARGS ( args , parse ) ;
2005-10-19 18:19:02 +00:00
2009-03-12 13:24:12 +00:00
if ( ! ast_strlen_zero ( args . sleep ) & & ( sleepms = atoi ( args . sleep ) ) )
2008-08-10 14:45:25 +00:00
sleepms * = 1000 ;
2007-07-23 19:51:41 +00:00
2009-03-12 13:24:12 +00:00
if ( ! ast_strlen_zero ( args . retries ) ) {
loops = atoi ( args . retries ) ;
}
2007-07-23 19:51:41 +00:00
if ( ! args . dialdata ) {
ast_log ( LOG_ERROR , " %s requires a 4th argument (dialdata) \n " , rapp ) ;
2006-04-19 14:02:49 +00:00
goto done ;
2005-01-18 03:12:53 +00:00
}
2008-02-09 11:27:10 +00:00
2008-08-10 14:45:25 +00:00
if ( sleepms < 1000 )
sleepms = 10000 ;
2006-04-19 14:02:49 +00:00
2005-01-18 03:12:53 +00:00
if ( ! loops )
2008-02-09 11:27:10 +00:00
loops = - 1 ; /* run forever */
2008-06-18 13:09:02 +00:00
ast_channel_lock ( chan ) ;
2005-01-18 03:12:53 +00:00
context = pbx_builtin_getvar_helper ( chan , " EXITCONTEXT " ) ;
2008-06-18 13:09:02 +00:00
context = ! ast_strlen_zero ( context ) ? ast_strdupa ( context ) : NULL ;
ast_channel_unlock ( chan ) ;
2006-04-19 14:02:49 +00:00
res = 0 ;
2005-01-18 03:12:53 +00:00
while ( loops ) {
2007-02-15 16:24:13 +00:00
int continue_exec ;
2005-01-18 03:12:53 +00:00
chan - > data = " Retrying " ;
if ( ast_test_flag ( chan , AST_FLAG_MOH ) )
ast_moh_stop ( chan ) ;
2007-07-23 19:51:41 +00:00
res = dial_exec_full ( chan , args . dialdata , & peerflags , & continue_exec ) ;
2007-02-15 16:24:13 +00:00
if ( continue_exec )
2007-01-10 04:56:48 +00:00
break ;
2007-07-16 18:18:19 +00:00
2007-02-15 16:24:13 +00:00
if ( res = = 0 ) {
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 23:24:27 +00:00
if ( ast_test_flag64 ( & peerflags , OPT_DTMF_EXIT ) ) {
2007-07-23 19:51:41 +00:00
if ( ! ast_strlen_zero ( args . announce ) ) {
if ( ast_fileexists ( args . announce , NULL , chan - > language ) > 0 ) {
2008-02-09 11:27:10 +00:00
if ( ! ( res = ast_streamfile ( chan , args . announce , chan - > language ) ) )
2007-07-16 18:18:19 +00:00
ast_waitstream ( chan , AST_DIGIT_ANY ) ;
} else
2007-07-23 19:51:41 +00:00
ast_log ( LOG_WARNING , " Announce file \" %s \" specified in Retrydial does not exist \n " , args . announce ) ;
2007-07-16 18:18:19 +00:00
}
2008-08-10 14:45:25 +00:00
if ( ! res & & sleepms ) {
2005-01-18 03:12:53 +00:00
if ( ! ast_test_flag ( chan , AST_FLAG_MOH ) )
2006-07-19 20:44:39 +00:00
ast_moh_start ( chan , NULL , NULL ) ;
2008-08-10 14:45:25 +00:00
res = ast_waitfordigit ( chan , sleepms ) ;
2005-01-18 03:12:53 +00:00
}
} else {
2007-07-23 19:51:41 +00:00
if ( ! ast_strlen_zero ( args . announce ) ) {
if ( ast_fileexists ( args . announce , NULL , chan - > language ) > 0 ) {
if ( ! ( res = ast_streamfile ( chan , args . announce , chan - > language ) ) )
2007-07-16 18:18:19 +00:00
res = ast_waitstream ( chan , " " ) ;
} else
2007-07-23 19:51:41 +00:00
ast_log ( LOG_WARNING , " Announce file \" %s \" specified in Retrydial does not exist \n " , args . announce ) ;
2007-07-16 18:18:19 +00:00
}
2008-08-10 14:45:25 +00:00
if ( sleepms ) {
2005-01-20 22:59:50 +00:00
if ( ! ast_test_flag ( chan , AST_FLAG_MOH ) )
2006-07-19 20:44:39 +00:00
ast_moh_start ( chan , NULL , NULL ) ;
2007-07-16 18:18:19 +00:00
if ( ! res )
2008-08-10 14:45:25 +00:00
res = ast_waitfordigit ( chan , sleepms ) ;
2005-01-20 22:59:50 +00:00
}
2005-01-18 03:12:53 +00:00
}
}
2008-04-28 16:37:45 +00:00
if ( res < 0 | | res = = AST_PBX_INCOMPLETE ) {
2005-01-18 03:12:53 +00:00
break ;
2008-04-28 16:37:45 +00:00
} else if ( res > 0 ) { /* Trying to send the call elsewhere (1 digit ext) */
2005-04-29 15:04:26 +00:00
if ( onedigit_goto ( chan , context , ( char ) res , 1 ) ) {
2005-01-18 03:12:53 +00:00
res = 0 ;
break ;
}
}
loops - - ;
}
2006-04-19 14:02:49 +00:00
if ( loops = = 0 )
res = 0 ;
2007-01-10 04:56:48 +00:00
else if ( res = = 1 )
res = 0 ;
2005-01-18 03:12:53 +00:00
if ( ast_test_flag ( chan , AST_FLAG_MOH ) )
ast_moh_stop ( chan ) ;
2007-01-10 04:56:48 +00:00
done :
2006-04-19 14:02:49 +00:00
return res ;
2005-01-18 03:12:53 +00:00
}
2006-08-21 02:11:39 +00:00
static int unload_module ( void )
1999-12-04 21:35:07 +00:00
{
2005-10-18 22:52:21 +00:00
int res ;
2007-06-20 17:35:08 +00:00
struct ast_context * con ;
2005-10-18 22:52:21 +00:00
res = ast_unregister_application ( app ) ;
res | = ast_unregister_application ( rapp ) ;
2008-02-09 11:27:10 +00:00
if ( ( con = ast_context_find ( " app_dial_gosub_virtual_context " ) ) ) {
2008-06-16 20:43:46 +00:00
ast_context_remove_extension2 ( con , " s " , 1 , NULL , 0 ) ;
closes issue #11285, where an unload of a module that creates a dialplan context, causes a crash when you do a 'dialplan show' of that context. This is because the registrar string is defined in the module, and the stale pointer is traversed. The reporter offered a patch that would always strdup the registrar string, which is practical, but I preferred to destroy the created contexts in each module where one is created. That seemed more symmetric. There were only 6 place in asterisk where this is done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, and app_queue. The two apps destroyed the context, but left the contexts. All is fixed now and unloads should be dialplan friendly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21 23:54:12 +00:00
ast_context_destroy ( con , " app_dial " ) ; /* leave nothing behind */
}
2008-02-09 11:27:10 +00:00
2005-10-18 22:52:21 +00:00
return res ;
1999-12-04 21:35:07 +00:00
}
2006-08-21 02:11:39 +00:00
static int load_module ( void )
1999-12-04 21:35:07 +00:00
{
2000-03-26 01:59:06 +00:00
int res ;
2007-06-20 17:35:08 +00:00
struct ast_context * con ;
(closes issue #6002)
Reported by: rizzo
Tested by: murf
Proposal of the changes to be made, and then an announcement of how they were accomplished:
http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html
and:
http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html
Here is a recap, file by file, of what I have done:
pbx/pbx_config.c
pbx/pbx_ael.c
All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.
We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.
pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and
then call merge_contexts_and_delete, which will merge (now) existing contexts and
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then
destroy the old dialplan.
chan_sip.c
chan_iax.c
chan_skinny.c
All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.
chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.
apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c
All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.
include/asterisk/pbx.h
ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create() interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.
include/asterisk/pval.h
ast_compile_ael2() interface changed to include the local hashtab table ptr.
main/features.c
For the sake of the parking context, we use ast_context_find_or_create().
main/pbx.c
I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.
refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.
Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.
Added some calls to ast_verb(3,...) for debug messages
Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.
find_or_create was upgraded to handle both local lists/tables as well as the globals.
context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables
ast_merge_contexts_and_delete() was heavily modified.
ast_add_extension2() was also upgraded to handle changes.
the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.
res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile
Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps. The main gotcha was I had to
include lock.h and hashtab.h in several places.
As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.
How's this for verbose commit messages?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 18:57:57 +00:00
con = ast_context_find_or_create ( NULL , NULL , " app_dial_gosub_virtual_context " , " app_dial " ) ;
2007-06-20 17:35:08 +00:00
if ( ! con )
ast_log ( LOG_ERROR , " Dial virtual context 'app_dial_gosub_virtual_context' does not exist and unable to create \n " ) ;
else
2008-12-18 19:33:42 +00:00
ast_add_extension2 ( con , 1 , " s " , 1 , NULL , NULL , " NoOp " , ast_strdup ( " " ) , ast_free_ptr , " app_dial " ) ;
2005-10-18 22:52:21 +00:00
2008-11-01 21:10:07 +00:00
res = ast_register_application_xml ( app , dial_exec ) ;
res | = ast_register_application_xml ( rapp , retrydial_exec ) ;
2008-02-09 11:27:10 +00:00
2000-03-26 01:59:06 +00:00
return res ;
1999-12-04 21:35:07 +00:00
}
2006-08-21 02:11:39 +00:00
AST_MODULE_INFO_STANDARD ( ASTERISK_GPL_KEY , " Dialing Application " ) ;