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https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67066 | file | 2007-06-04 13:59:14 -0400 (Mon, 04 Jun 2007) | 2 lines Initialize cidname variable to nothing since it may be used without having been touched. (issue #9661 reported by dimas) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1875 lines
67 KiB
C
1875 lines
67 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2006, Digium, Inc.
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*
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* Mark Spencer <markster@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief dial() & retrydial() - Trivial application to dial a channel and send an URL on answer
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*
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* \author Mark Spencer <markster@digium.com>
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*
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* \ingroup applications
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*/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <stdlib.h>
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#include <errno.h>
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#include <unistd.h>
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#include <string.h>
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#include <stdlib.h>
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#include <stdio.h>
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#include <sys/time.h>
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#include <sys/signal.h>
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#include <sys/stat.h>
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#include <netinet/in.h>
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#include "asterisk/lock.h"
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#include "asterisk/file.h"
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#include "asterisk/logger.h"
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#include "asterisk/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/options.h"
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#include "asterisk/module.h"
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#include "asterisk/translate.h"
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#include "asterisk/say.h"
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#include "asterisk/config.h"
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#include "asterisk/features.h"
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#include "asterisk/musiconhold.h"
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#include "asterisk/callerid.h"
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#include "asterisk/utils.h"
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#include "asterisk/app.h"
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#include "asterisk/causes.h"
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#include "asterisk/rtp.h"
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#include "asterisk/manager.h"
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#include "asterisk/privacy.h"
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#include "asterisk/stringfields.h"
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static char *app = "Dial";
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static char *synopsis = "Place a call and connect to the current channel";
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static char *descrip =
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" Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]):\n"
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"This application will place calls to one or more specified channels. As soon\n"
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"as one of the requested channels answers, the originating channel will be\n"
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"answered, if it has not already been answered. These two channels will then\n"
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"be active in a bridged call. All other channels that were requested will then\n"
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"be hung up.\n"
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" Unless there is a timeout specified, the Dial application will wait\n"
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"indefinitely until one of the called channels answers, the user hangs up, or\n"
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"if all of the called channels are busy or unavailable. Dialplan executing will\n"
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"continue if no requested channels can be called, or if the timeout expires.\n\n"
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" This application sets the following channel variables upon completion:\n"
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" DIALEDTIME - This is the time from dialing a channel until when it\n"
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" is disconnected.\n"
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" ANSWEREDTIME - This is the amount of time for actual call.\n"
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" DIALSTATUS - This is the status of the call:\n"
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" CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n"
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" DONTCALL | TORTURE | INVALIDARGS\n"
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" For the Privacy and Screening Modes, the DIALSTATUS variable will be set to\n"
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"DONTCALL if the called party chooses to send the calling party to the 'Go Away'\n"
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"script. The DIALSTATUS variable will be set to TORTURE if the called party\n"
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"wants to send the caller to the 'torture' script.\n"
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" This application will report normal termination if the originating channel\n"
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"hangs up, or if the call is bridged and either of the parties in the bridge\n"
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"ends the call.\n"
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" The optional URL will be sent to the called party if the channel supports it.\n"
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" If the OUTBOUND_GROUP variable is set, all peer channels created by this\n"
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"application will be put into that group (as in Set(GROUP()=...).\n"
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" If the OUTBOUND_GROUP_ONCE variable is set, all peer channels created by this\n"
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"application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,\n"
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"however, the variable will be unset after use.\n\n"
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" Options:\n"
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" A(x) - Play an announcement to the called party, using 'x' as the file.\n"
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" C - Reset the CDR for this call.\n"
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" d - Allow the calling user to dial a 1 digit extension while waiting for\n"
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" a call to be answered. Exit to that extension if it exists in the\n"
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" current context, or the context defined in the EXITCONTEXT variable,\n"
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" if it exists.\n"
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" D([called][:calling]) - Send the specified DTMF strings *after* the called\n"
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" party has answered, but before the call gets bridged. The 'called'\n"
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" DTMF string is sent to the called party, and the 'calling' DTMF\n"
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" string is sent to the calling party. Both parameters can be used\n"
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" alone.\n"
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" f - Force the callerid of the *calling* channel to be set as the\n"
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" extension associated with the channel using a dialplan 'hint'.\n"
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" For example, some PSTNs do not allow CallerID to be set to anything\n"
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" other than the number assigned to the caller.\n"
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" g - Proceed with dialplan execution at the current extension if the\n"
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" destination channel hangs up.\n"
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" G(context^exten^pri) - If the call is answered, transfer the calling party to\n"
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" the specified priority and the called party to the specified priority+1.\n"
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" Optionally, an extension, or extension and context may be specified. \n"
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" Otherwise, the current extension is used. You cannot use any additional\n"
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" action post answer options in conjunction with this option.\n"
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" h - Allow the called party to hang up by sending the '*' DTMF digit.\n"
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" H - Allow the calling party to hang up by hitting the '*' DTMF digit.\n"
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" i - Asterisk will ignore any forwarding requests it may receive on this\n"
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" dial attempt.\n"
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" j - Jump to priority n+101 if all of the requested channels were busy.\n"
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" L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are\n"
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" left. Repeat the warning every 'z' ms. The following special\n"
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" variables can be used with this option:\n"
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" * LIMIT_PLAYAUDIO_CALLER yes|no (default yes)\n"
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" Play sounds to the caller.\n"
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" * LIMIT_PLAYAUDIO_CALLEE yes|no\n"
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" Play sounds to the callee.\n"
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" * LIMIT_TIMEOUT_FILE File to play when time is up.\n"
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" * LIMIT_CONNECT_FILE File to play when call begins.\n"
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" * LIMIT_WARNING_FILE File to play as warning if 'y' is defined.\n"
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" The default is to say the time remaining.\n"
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" m([class]) - Provide hold music to the calling party until a requested\n"
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" channel answers. A specific MusicOnHold class can be\n"
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" specified.\n"
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" M(x[^arg]) - Execute the Macro for the *called* channel before connecting\n"
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" to the calling channel. Arguments can be specified to the Macro\n"
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" using '^' as a delimeter. The Macro can set the variable\n"
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" MACRO_RESULT to specify the following actions after the Macro is\n"
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" finished executing.\n"
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" * ABORT Hangup both legs of the call.\n"
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" * CONGESTION Behave as if line congestion was encountered.\n"
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" * BUSY Behave as if a busy signal was encountered. This will also\n"
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" have the application jump to priority n+101 if the\n"
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" 'j' option is set.\n"
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" * CONTINUE Hangup the called party and allow the calling party\n"
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" to continue dialplan execution at the next priority.\n"
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" * GOTO:<context>^<exten>^<priority> - Transfer the call to the\n"
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" specified priority. Optionally, an extension, or\n"
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" extension and priority can be specified.\n"
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" You cannot use any additional action post answer options in conjunction\n"
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" with this option. Also, pbx services are not run on the peer (called) channel,\n"
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" so you will not be able to set timeouts via the TIMEOUT() function in this macro.\n"
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" n - This option is a modifier for the screen/privacy mode. It specifies\n"
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" that no introductions are to be saved in the priv-callerintros\n"
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" directory.\n"
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" N - This option is a modifier for the screen/privacy mode. It specifies\n"
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" that if callerID is present, do not screen the call.\n"
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" o - Specify that the CallerID that was present on the *calling* channel\n"
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" be set as the CallerID on the *called* channel. This was the\n"
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" behavior of Asterisk 1.0 and earlier.\n"
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" O([x]) - \"Operator Services\" mode (Zaptel channel to Zaptel channel\n"
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" only, if specified on non-Zaptel interface, it will be ignored).\n"
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" When the destination answers (presumably an operator services\n"
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" station), the originator no longer has control of their line.\n"
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" They may hang up, but the switch will not release their line\n"
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" until the destination party hangs up (the operator). Specified\n"
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" without an arg, or with 1 as an arg, the originator hanging up\n"
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" will cause the phone to ring back immediately. With a 2 specified,\n"
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" when the \"operator\" flashes the trunk, it will ring their phone\n"
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" back.\n"
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" p - This option enables screening mode. This is basically Privacy mode\n"
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" without memory.\n"
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" P([x]) - Enable privacy mode. Use 'x' as the family/key in the database if\n"
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" it is provided. The current extension is used if a database\n"
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" family/key is not specified.\n"
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" r - Indicate ringing to the calling party. Pass no audio to the calling\n"
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" party until the called channel has answered.\n"
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" S(x) - Hang up the call after 'x' seconds *after* the called party has\n"
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" answered the call.\n"
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" t - Allow the called party to transfer the calling party by sending the\n"
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" DTMF sequence defined in features.conf.\n"
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" T - Allow the calling party to transfer the called party by sending the\n"
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" DTMF sequence defined in features.conf.\n"
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" w - Allow the called party to enable recording of the call by sending\n"
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" the DTMF sequence defined for one-touch recording in features.conf.\n"
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" W - Allow the calling party to enable recording of the call by sending\n"
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" the DTMF sequence defined for one-touch recording in features.conf.\n"
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" k - Allow the called party to enable parking of the call by sending\n"
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" the DTMF sequence defined for call parking in features.conf.\n"
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" K - Allow the calling party to enable parking of the call by sending\n"
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" the DTMF sequence defined for call parking in features.conf.\n";
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/* RetryDial App by Anthony Minessale II <anthmct@yahoo.com> Jan/2005 */
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static char *rapp = "RetryDial";
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static char *rsynopsis = "Place a call, retrying on failure allowing optional exit extension.";
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static char *rdescrip =
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" RetryDial(announce|sleep|retries|dialargs): This application will attempt to\n"
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"place a call using the normal Dial application. If no channel can be reached,\n"
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"the 'announce' file will be played. Then, it will wait 'sleep' number of\n"
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"seconds before retying the call. After 'retires' number of attempts, the\n"
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"calling channel will continue at the next priority in the dialplan. If the\n"
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"'retries' setting is set to 0, this application will retry endlessly.\n"
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" While waiting to retry a call, a 1 digit extension may be dialed. If that\n"
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"extension exists in either the context defined in ${EXITCONTEXT} or the current\n"
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"one, The call will jump to that extension immediately.\n"
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" The 'dialargs' are specified in the same format that arguments are provided\n"
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"to the Dial application.\n";
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enum {
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OPT_ANNOUNCE = (1 << 0),
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OPT_RESETCDR = (1 << 1),
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OPT_DTMF_EXIT = (1 << 2),
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OPT_SENDDTMF = (1 << 3),
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OPT_FORCECLID = (1 << 4),
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OPT_GO_ON = (1 << 5),
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OPT_CALLEE_HANGUP = (1 << 6),
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OPT_CALLER_HANGUP = (1 << 7),
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OPT_PRIORITY_JUMP = (1 << 8),
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OPT_DURATION_LIMIT = (1 << 9),
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OPT_MUSICBACK = (1 << 10),
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OPT_CALLEE_MACRO = (1 << 11),
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OPT_SCREEN_NOINTRO = (1 << 12),
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OPT_SCREEN_NOCLID = (1 << 13),
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OPT_ORIGINAL_CLID = (1 << 14),
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OPT_SCREENING = (1 << 15),
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OPT_PRIVACY = (1 << 16),
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OPT_RINGBACK = (1 << 17),
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OPT_DURATION_STOP = (1 << 18),
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OPT_CALLEE_TRANSFER = (1 << 19),
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OPT_CALLER_TRANSFER = (1 << 20),
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OPT_CALLEE_MONITOR = (1 << 21),
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OPT_CALLER_MONITOR = (1 << 22),
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OPT_GOTO = (1 << 23),
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OPT_OPERMODE = (1 << 24),
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OPT_CALLEE_PARK = (1 << 25),
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OPT_CALLER_PARK = (1 << 26),
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OPT_IGNORE_FORWARDING = (1 << 27),
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};
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#define DIAL_STILLGOING (1 << 30)
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#define DIAL_NOFORWARDHTML (1 << 31)
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enum {
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OPT_ARG_ANNOUNCE = 0,
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OPT_ARG_SENDDTMF,
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OPT_ARG_GOTO,
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OPT_ARG_DURATION_LIMIT,
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OPT_ARG_MUSICBACK,
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OPT_ARG_CALLEE_MACRO,
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OPT_ARG_PRIVACY,
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OPT_ARG_DURATION_STOP,
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OPT_ARG_OPERMODE,
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/* note: this entry _MUST_ be the last one in the enum */
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OPT_ARG_ARRAY_SIZE,
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};
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AST_APP_OPTIONS(dial_exec_options, {
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AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
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AST_APP_OPTION('C', OPT_RESETCDR),
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AST_APP_OPTION('d', OPT_DTMF_EXIT),
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AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
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AST_APP_OPTION('f', OPT_FORCECLID),
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AST_APP_OPTION('g', OPT_GO_ON),
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AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
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AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
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AST_APP_OPTION('H', OPT_CALLER_HANGUP),
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AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
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AST_APP_OPTION('j', OPT_PRIORITY_JUMP),
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AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
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AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
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AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
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AST_APP_OPTION('n', OPT_SCREEN_NOINTRO),
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AST_APP_OPTION('N', OPT_SCREEN_NOCLID),
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AST_APP_OPTION_ARG('O', OPT_OPERMODE,OPT_ARG_OPERMODE),
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AST_APP_OPTION('o', OPT_ORIGINAL_CLID),
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AST_APP_OPTION('p', OPT_SCREENING),
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AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
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AST_APP_OPTION('r', OPT_RINGBACK),
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AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
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AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
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AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
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AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
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AST_APP_OPTION('W', OPT_CALLER_MONITOR),
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AST_APP_OPTION('k', OPT_CALLEE_PARK),
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AST_APP_OPTION('K', OPT_CALLER_PARK),
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});
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/*
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* The list of active channels
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*/
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struct chanlist {
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struct chanlist *next;
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struct ast_channel *chan;
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unsigned int flags;
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int forwards;
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};
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static void hanguptree(struct chanlist *outgoing, struct ast_channel *exception)
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{
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/* Hang up a tree of stuff */
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struct chanlist *oo;
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while (outgoing) {
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/* Hangup any existing lines we have open */
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if (outgoing->chan && (outgoing->chan != exception))
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ast_hangup(outgoing->chan);
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oo = outgoing;
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outgoing=outgoing->next;
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free(oo);
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}
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}
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#define AST_MAX_FORWARDS 8
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#define AST_MAX_WATCHERS 256
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/*
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* argument to handle_cause() and other functions.
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*/
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struct cause_args {
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struct ast_channel *chan;
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int busy;
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int congestion;
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int nochan;
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};
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static void handle_cause(int cause, struct cause_args *num)
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{
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struct ast_cdr *cdr = num->chan->cdr;
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switch(cause) {
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case AST_CAUSE_BUSY:
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if (cdr)
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ast_cdr_busy(cdr);
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num->busy++;
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break;
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case AST_CAUSE_CONGESTION:
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if (cdr)
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ast_cdr_failed(cdr);
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num->congestion++;
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break;
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case AST_CAUSE_UNREGISTERED:
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if (cdr)
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ast_cdr_failed(cdr);
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num->nochan++;
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break;
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case AST_CAUSE_NORMAL_CLEARING:
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break;
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default:
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num->nochan++;
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break;
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}
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}
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/* free the buffer if allocated, and set the pointer to the second arg */
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#define S_REPLACE(s, new_val) \
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do { \
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if (s) \
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free(s); \
|
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s = (new_val); \
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} while (0)
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static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
|
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{
|
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char rexten[2] = { exten, '\0' };
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if (context) {
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if (!ast_goto_if_exists(chan, context, rexten, pri))
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return 1;
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} else {
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if (!ast_goto_if_exists(chan, chan->context, rexten, pri))
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return 1;
|
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else if (!ast_strlen_zero(chan->macrocontext)) {
|
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if (!ast_goto_if_exists(chan, chan->macrocontext, rexten, pri))
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return 1;
|
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}
|
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}
|
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return 0;
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}
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|
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static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
|
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{
|
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const char *context = S_OR(chan->macrocontext, chan->context);
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const char *exten = S_OR(chan->macroexten, chan->exten);
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return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
|
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}
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static void senddialevent(struct ast_channel *src, struct ast_channel *dst)
|
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{
|
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manager_event(EVENT_FLAG_CALL, "Dial",
|
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"SubEvent: Begin\r\n"
|
|
"Source: %s\r\n"
|
|
"Destination: %s\r\n"
|
|
"CallerIDNum: %s\r\n"
|
|
"CallerIDName: %s\r\n"
|
|
"SrcUniqueID: %s\r\n"
|
|
"DestUniqueID: %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);
|
|
}
|
|
|
|
static void senddialendevent(const struct ast_channel *src, const char *dialstatus)
|
|
{
|
|
manager_event(EVENT_FLAG_CALL, "Dial",
|
|
"SubEvent: End\r\n"
|
|
"Channel: %s\r\n"
|
|
"DialStatus: %s\r\n",
|
|
src->name, dialstatus);
|
|
}
|
|
|
|
/*!
|
|
* helper function for wait_for_answer()
|
|
*
|
|
* XXX this code is highly suspicious, as it essentially overwrites
|
|
* the outgoing channel without properly deleting it.
|
|
*/
|
|
static void do_forward(struct chanlist *o,
|
|
struct cause_args *num, struct ast_flags *peerflags, int single)
|
|
{
|
|
char tmpchan[256];
|
|
struct ast_channel *original = o->chan;
|
|
struct ast_channel *c = o->chan; /* the winner */
|
|
struct ast_channel *in = num->chan; /* the input channel */
|
|
char *stuff;
|
|
char *tech;
|
|
int cause;
|
|
|
|
ast_copy_string(tmpchan, c->call_forward, sizeof(tmpchan));
|
|
if ((stuff = strchr(tmpchan, '/'))) {
|
|
*stuff++ = '\0';
|
|
tech = tmpchan;
|
|
} else {
|
|
const char *forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
|
|
snprintf(tmpchan, sizeof(tmpchan), "%s@%s", c->call_forward, forward_context ? forward_context : c->context);
|
|
stuff = tmpchan;
|
|
tech = "Local";
|
|
}
|
|
/* Before processing channel, go ahead and check for forwarding */
|
|
o->forwards++;
|
|
if (o->forwards < AST_MAX_FORWARDS) {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_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_flag(peerflags, OPT_IGNORE_FORWARDING)) {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Forwarding %s to '%s/%s' prevented.\n", in->name, tech, stuff);
|
|
c = o->chan = NULL;
|
|
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);
|
|
} else
|
|
ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, cause);
|
|
}
|
|
} else {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Too many forwards from %s\n", c->name);
|
|
cause = AST_CAUSE_CONGESTION;
|
|
c = o->chan = NULL;
|
|
}
|
|
if (!c) {
|
|
ast_clear_flag(o, DIAL_STILLGOING);
|
|
handle_cause(cause, num);
|
|
} else {
|
|
char *new_cid_num, *new_cid_name;
|
|
struct ast_channel *src;
|
|
|
|
ast_rtp_make_compatible(c, in, single);
|
|
if (ast_test_flag(o, OPT_FORCECLID)) {
|
|
new_cid_num = ast_strdup(S_OR(in->macroexten, in->exten));
|
|
new_cid_name = NULL; /* XXX no name ? */
|
|
src = c; /* XXX possible bug in previous code, which used 'winner' ? it may have changed */
|
|
} else {
|
|
new_cid_num = ast_strdup(in->cid.cid_num);
|
|
new_cid_name = ast_strdup(in->cid.cid_name);
|
|
src = in;
|
|
}
|
|
ast_string_field_set(c, accountcode, src->accountcode);
|
|
c->cdrflags = src->cdrflags;
|
|
S_REPLACE(c->cid.cid_num, new_cid_num);
|
|
S_REPLACE(c->cid.cid_name, new_cid_name);
|
|
|
|
if (in->cid.cid_ani) { /* XXX or maybe unconditional ? */
|
|
S_REPLACE(c->cid.cid_ani, ast_strdup(in->cid.cid_ani));
|
|
}
|
|
S_REPLACE(c->cid.cid_rdnis, ast_strdup(S_OR(in->macroexten, in->exten)));
|
|
if (ast_call(c, tmpchan, 0)) {
|
|
ast_log(LOG_NOTICE, "Failed to dial on local channel for call forward to '%s'\n", tmpchan);
|
|
ast_clear_flag(o, DIAL_STILLGOING);
|
|
ast_hangup(original);
|
|
c = o->chan = NULL;
|
|
num->nochan++;
|
|
} else {
|
|
senddialevent(in, c);
|
|
/* After calling, set callerid to extension */
|
|
if (!ast_test_flag(peerflags, OPT_ORIGINAL_CLID)) {
|
|
char cidname[AST_MAX_EXTENSION] = "";
|
|
ast_set_callerid(c, S_OR(in->macroexten, in->exten), get_cid_name(cidname, sizeof(cidname), in), NULL);
|
|
}
|
|
/* Hangup the original channel now, in case we needed it */
|
|
ast_hangup(original);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* argument used for some functions. */
|
|
struct privacy_args {
|
|
int sentringing;
|
|
int privdb_val;
|
|
char privcid[256];
|
|
char privintro[1024];
|
|
char status[256];
|
|
};
|
|
|
|
static struct ast_channel *wait_for_answer(struct ast_channel *in,
|
|
struct chanlist *outgoing, int *to, struct ast_flags *peerflags,
|
|
struct privacy_args *pa,
|
|
const struct cause_args *num_in, int priority_jump, int *result)
|
|
{
|
|
struct cause_args num = *num_in;
|
|
int prestart = num.busy + num.congestion + num.nochan;
|
|
int orig = *to;
|
|
struct ast_channel *peer = NULL;
|
|
/* single is set if only one destination is enabled */
|
|
int single = outgoing && !outgoing->next && !ast_test_flag(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
|
|
|
|
if (single) {
|
|
/* Turn off hold music, etc */
|
|
ast_deactivate_generator(in);
|
|
/* If we are calling a single channel, make them compatible for in-band tone purpose */
|
|
ast_channel_make_compatible(outgoing->chan, in);
|
|
}
|
|
|
|
|
|
while (*to && !peer) {
|
|
struct chanlist *o;
|
|
int pos = 0; /* how many channels do we handle */
|
|
int numlines = prestart;
|
|
struct ast_channel *winner;
|
|
struct ast_channel *watchers[AST_MAX_WATCHERS];
|
|
|
|
watchers[pos++] = in;
|
|
for (o = outgoing; o; o = o->next) {
|
|
/* Keep track of important channels */
|
|
if (ast_test_flag(o, DIAL_STILLGOING) && o->chan)
|
|
watchers[pos++] = o->chan;
|
|
numlines++;
|
|
}
|
|
if (pos == 1) { /* only the input channel is available */
|
|
if (numlines == (num.busy + num.congestion + num.nochan)) {
|
|
if (option_verbose > 2)
|
|
ast_verbose( VERBOSE_PREFIX_2 "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
|
|
if (num.busy)
|
|
strcpy(pa->status, "BUSY");
|
|
else if (num.congestion)
|
|
strcpy(pa->status, "CONGESTION");
|
|
else if (num.nochan)
|
|
strcpy(pa->status, "CHANUNAVAIL");
|
|
if (ast_opt_priority_jumping || priority_jump)
|
|
ast_goto_if_exists(in, in->context, in->exten, in->priority + 101);
|
|
} else {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
|
|
}
|
|
*to = 0;
|
|
return NULL;
|
|
}
|
|
winner = ast_waitfor_n(watchers, pos, to);
|
|
for (o = outgoing; o; o = o->next) {
|
|
struct ast_frame *f;
|
|
struct ast_channel *c = o->chan;
|
|
|
|
if (c == NULL)
|
|
continue;
|
|
if (ast_test_flag(o, DIAL_STILLGOING) && c->_state == AST_STATE_UP) {
|
|
if (!peer) {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "%s answered %s\n", c->name, in->name);
|
|
peer = c;
|
|
ast_copy_flags(peerflags, o,
|
|
OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
|
|
OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
|
|
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
|
|
OPT_CALLEE_PARK | OPT_CALLER_PARK |
|
|
DIAL_NOFORWARDHTML);
|
|
ast_copy_string(c->dialcontext, "", sizeof(c->dialcontext));
|
|
ast_copy_string(c->exten, "", sizeof(c->exten));
|
|
}
|
|
continue;
|
|
}
|
|
if (c != winner)
|
|
continue;
|
|
/* here, o->chan == c == winner */
|
|
if (!ast_strlen_zero(c->call_forward)) {
|
|
do_forward(o, &num, peerflags, single);
|
|
continue;
|
|
}
|
|
f = ast_read(winner);
|
|
if (!f) {
|
|
in->hangupcause = c->hangupcause;
|
|
ast_hangup(c);
|
|
c = o->chan = NULL;
|
|
ast_clear_flag(o, DIAL_STILLGOING);
|
|
handle_cause(in->hangupcause, &num);
|
|
continue;
|
|
}
|
|
if (f->frametype == AST_FRAME_CONTROL) {
|
|
switch(f->subclass) {
|
|
case AST_CONTROL_ANSWER:
|
|
/* This is our guy if someone answered. */
|
|
if (!peer) {
|
|
if (option_verbose > 2)
|
|
ast_verbose( VERBOSE_PREFIX_3 "%s answered %s\n", c->name, in->name);
|
|
peer = c;
|
|
ast_copy_flags(peerflags, o,
|
|
OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
|
|
OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
|
|
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
|
|
OPT_CALLEE_PARK | OPT_CALLER_PARK |
|
|
DIAL_NOFORWARDHTML);
|
|
ast_copy_string(c->dialcontext, "", sizeof(c->dialcontext));
|
|
ast_copy_string(c->exten, "", sizeof(c->exten));
|
|
/* Setup early bridge if appropriate */
|
|
ast_channel_early_bridge(in, peer);
|
|
}
|
|
/* 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:
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "%s is busy\n", c->name);
|
|
in->hangupcause = c->hangupcause;
|
|
ast_hangup(c);
|
|
c = o->chan = NULL;
|
|
ast_clear_flag(o, DIAL_STILLGOING);
|
|
handle_cause(AST_CAUSE_BUSY, &num);
|
|
break;
|
|
case AST_CONTROL_CONGESTION:
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "%s is circuit-busy\n", c->name);
|
|
in->hangupcause = c->hangupcause;
|
|
ast_hangup(c);
|
|
c = o->chan = NULL;
|
|
ast_clear_flag(o, DIAL_STILLGOING);
|
|
handle_cause(AST_CAUSE_CONGESTION, &num);
|
|
break;
|
|
case AST_CONTROL_RINGING:
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "%s is ringing\n", c->name);
|
|
/* Setup early media if appropriate */
|
|
if (single)
|
|
ast_channel_early_bridge(in, c);
|
|
if (!(pa->sentringing) && !ast_test_flag(outgoing, OPT_MUSICBACK)) {
|
|
ast_indicate(in, AST_CONTROL_RINGING);
|
|
pa->sentringing++;
|
|
}
|
|
break;
|
|
case AST_CONTROL_PROGRESS:
|
|
if (option_verbose > 2)
|
|
ast_verbose (VERBOSE_PREFIX_3 "%s is making progress passing it to %s\n", c->name, in->name);
|
|
/* Setup early media if appropriate */
|
|
if (single)
|
|
ast_channel_early_bridge(in, c);
|
|
if (!ast_test_flag(outgoing, OPT_RINGBACK))
|
|
ast_indicate(in, AST_CONTROL_PROGRESS);
|
|
break;
|
|
case AST_CONTROL_VIDUPDATE:
|
|
if (option_verbose > 2)
|
|
ast_verbose (VERBOSE_PREFIX_3 "%s requested a video update, passing it to %s\n", c->name, in->name);
|
|
ast_indicate(in, AST_CONTROL_VIDUPDATE);
|
|
break;
|
|
case AST_CONTROL_PROCEEDING:
|
|
if (option_verbose > 2)
|
|
ast_verbose (VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", c->name, in->name);
|
|
if (single)
|
|
ast_channel_early_bridge(in, c);
|
|
if (!ast_test_flag(outgoing, OPT_RINGBACK))
|
|
ast_indicate(in, AST_CONTROL_PROCEEDING);
|
|
break;
|
|
case AST_CONTROL_HOLD:
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Call on %s placed on hold\n", c->name);
|
|
ast_indicate(in, AST_CONTROL_HOLD);
|
|
break;
|
|
case AST_CONTROL_UNHOLD:
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Call on %s left from hold\n", c->name);
|
|
ast_indicate(in, AST_CONTROL_UNHOLD);
|
|
break;
|
|
case AST_CONTROL_OFFHOOK:
|
|
case AST_CONTROL_FLASH:
|
|
/* Ignore going off hook and flash */
|
|
break;
|
|
case -1:
|
|
if (!ast_test_flag(outgoing, OPT_RINGBACK | OPT_MUSICBACK)) {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "%s stopped sounds\n", c->name);
|
|
ast_indicate(in, -1);
|
|
pa->sentringing = 0;
|
|
}
|
|
break;
|
|
default:
|
|
if (option_debug)
|
|
ast_log(LOG_DEBUG, "Dunno what to do with control type %d\n", f->subclass);
|
|
}
|
|
} else if (single) {
|
|
/* XXX are we sure the logic is correct ? or we should just switch on f->frametype ? */
|
|
if (f->frametype == AST_FRAME_VOICE && !ast_test_flag(outgoing, OPT_RINGBACK|OPT_MUSICBACK)) {
|
|
if (ast_write(in, f))
|
|
ast_log(LOG_WARNING, "Unable to forward voice frame\n");
|
|
} else if (f->frametype == AST_FRAME_IMAGE && !ast_test_flag(outgoing, OPT_RINGBACK|OPT_MUSICBACK)) {
|
|
if (ast_write(in, f))
|
|
ast_log(LOG_WARNING, "Unable to forward image\n");
|
|
} else if (f->frametype == AST_FRAME_TEXT && !ast_test_flag(outgoing, OPT_RINGBACK|OPT_MUSICBACK)) {
|
|
if (ast_write(in, f))
|
|
ast_log(LOG_WARNING, "Unable to send text\n");
|
|
} else if (f->frametype == AST_FRAME_HTML && !ast_test_flag(outgoing, DIAL_NOFORWARDHTML)) {
|
|
if (ast_channel_sendhtml(in, f->subclass, f->data, f->datalen) == -1)
|
|
ast_log(LOG_WARNING, "Unable to send URL\n");
|
|
}
|
|
}
|
|
ast_frfree(f);
|
|
} /* end for */
|
|
if (winner == in) {
|
|
struct ast_frame *f = ast_read(in);
|
|
#if 0
|
|
if (f && (f->frametype != AST_FRAME_VOICE))
|
|
printf("Frame type: %d, %d\n", f->frametype, f->subclass);
|
|
else if (!f || (f->frametype != AST_FRAME_VOICE))
|
|
printf("Hangup received on %s\n", in->name);
|
|
#endif
|
|
if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP))) {
|
|
/* Got hung up */
|
|
*to = -1;
|
|
strcpy(pa->status, "CANCEL");
|
|
ast_cdr_noanswer(in->cdr);
|
|
if (f)
|
|
ast_frfree(f);
|
|
return NULL;
|
|
}
|
|
|
|
/* now f is guaranteed non-NULL */
|
|
if (f->frametype == AST_FRAME_DTMF) {
|
|
if (ast_test_flag(peerflags, OPT_DTMF_EXIT)) {
|
|
const char *context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
|
|
if (onedigit_goto(in, context, (char) f->subclass, 1)) {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
|
|
*to=0;
|
|
ast_cdr_noanswer(in->cdr);
|
|
*result = f->subclass;
|
|
strcpy(pa->status, "CANCEL");
|
|
ast_frfree(f);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
if (ast_test_flag(peerflags, OPT_CALLER_HANGUP) &&
|
|
(f->subclass == '*')) { /* hmm it it not guaranteed to be '*' anymore. */
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
|
|
*to=0;
|
|
strcpy(pa->status, "CANCEL");
|
|
ast_cdr_noanswer(in->cdr);
|
|
ast_frfree(f);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* Forward HTML stuff */
|
|
if (single && (f->frametype == AST_FRAME_HTML) && !ast_test_flag(outgoing, DIAL_NOFORWARDHTML))
|
|
if(ast_channel_sendhtml(outgoing->chan, f->subclass, f->data, f->datalen) == -1)
|
|
ast_log(LOG_WARNING, "Unable to send URL\n");
|
|
|
|
|
|
if (single && ((f->frametype == AST_FRAME_VOICE) || (f->frametype == AST_FRAME_DTMF_BEGIN) || (f->frametype == AST_FRAME_DTMF_END))) {
|
|
if (ast_write(outgoing->chan, f))
|
|
ast_log(LOG_WARNING, "Unable to forward voice or dtmf\n");
|
|
}
|
|
if (single && (f->frametype == AST_FRAME_CONTROL) &&
|
|
((f->subclass == AST_CONTROL_HOLD) ||
|
|
(f->subclass == AST_CONTROL_UNHOLD) ||
|
|
(f->subclass == AST_CONTROL_VIDUPDATE))) {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "%s requested special control %d, passing it to %s\n", in->name, f->subclass, outgoing->chan->name);
|
|
ast_indicate_data(outgoing->chan, f->subclass, f->data, f->datalen);
|
|
}
|
|
ast_frfree(f);
|
|
}
|
|
if (!*to && (option_verbose > 2))
|
|
ast_verbose(VERBOSE_PREFIX_3 "Nobody picked up in %d ms\n", orig);
|
|
if (!*to || ast_check_hangup(in)) {
|
|
ast_cdr_noanswer(in->cdr);
|
|
}
|
|
|
|
}
|
|
|
|
return peer;
|
|
}
|
|
|
|
static void replace_macro_delimiter(char *s)
|
|
{
|
|
for (; *s; s++)
|
|
if (*s == '^')
|
|
*s = '|';
|
|
}
|
|
|
|
|
|
/* returns true if there is a valid privacy reply */
|
|
static int valid_priv_reply(struct ast_flags *opts, int res)
|
|
{
|
|
if (res < '1')
|
|
return 0;
|
|
if (ast_test_flag(opts, OPT_PRIVACY) && res <= '5')
|
|
return 1;
|
|
if (ast_test_flag(opts, OPT_SCREENING) && res <= '4')
|
|
return 1;
|
|
return 0;
|
|
}
|
|
|
|
static int do_timelimit(struct ast_channel *chan, struct ast_bridge_config *config,
|
|
char *parse, unsigned int *calldurationlimit)
|
|
{
|
|
char *stringp = ast_strdupa(parse);
|
|
char *limit_str, *warning_str, *warnfreq_str;
|
|
const char *var;
|
|
int play_to_caller=0,play_to_callee=0;
|
|
int delta;
|
|
|
|
limit_str = strsep(&stringp, ":");
|
|
warning_str = strsep(&stringp, ":");
|
|
warnfreq_str = strsep(&stringp, ":");
|
|
|
|
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;
|
|
return -1; /* error */
|
|
} 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;
|
|
}
|
|
}
|
|
|
|
var = pbx_builtin_getvar_helper(chan,"LIMIT_PLAYAUDIO_CALLER");
|
|
play_to_caller = var ? ast_true(var) : 1;
|
|
|
|
var = pbx_builtin_getvar_helper(chan,"LIMIT_PLAYAUDIO_CALLEE");
|
|
play_to_callee = var ? ast_true(var) : 0;
|
|
|
|
if (!play_to_caller && !play_to_callee)
|
|
play_to_caller = 1;
|
|
|
|
var = pbx_builtin_getvar_helper(chan,"LIMIT_WARNING_FILE");
|
|
config->warning_sound = S_OR(var, "timeleft");
|
|
|
|
/* 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.
|
|
*/
|
|
var = pbx_builtin_getvar_helper(chan,"LIMIT_TIMEOUT_FILE");
|
|
config->end_sound = S_OR(var, NULL);
|
|
var = pbx_builtin_getvar_helper(chan,"LIMIT_CONNECT_FILE");
|
|
config->start_sound = S_OR(var, NULL);
|
|
|
|
/* undo effect of S(x) in case they are both used */
|
|
*calldurationlimit = 0;
|
|
/* 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) {
|
|
*calldurationlimit = config->timelimit / 1000;
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Setting call duration limit to %d seconds.\n",
|
|
*calldurationlimit);
|
|
config->timelimit = play_to_caller = play_to_callee =
|
|
config->play_warning = config->warning_freq = 0;
|
|
} else if (option_verbose > 2) {
|
|
ast_verbose(VERBOSE_PREFIX_3 "Limit Data for this call:\n");
|
|
ast_verbose(VERBOSE_PREFIX_4 "timelimit = %ld\n", config->timelimit);
|
|
ast_verbose(VERBOSE_PREFIX_4 "play_warning = %ld\n", config->play_warning);
|
|
ast_verbose(VERBOSE_PREFIX_4 "play_to_caller = %s\n", play_to_caller ? "yes" : "no");
|
|
ast_verbose(VERBOSE_PREFIX_4 "play_to_callee = %s\n", play_to_callee ? "yes" : "no");
|
|
ast_verbose(VERBOSE_PREFIX_4 "warning_freq = %ld\n", config->warning_freq);
|
|
ast_verbose(VERBOSE_PREFIX_4 "start_sound = %s\n", S_OR(config->start_sound, ""));
|
|
ast_verbose(VERBOSE_PREFIX_4 "warning_sound = %s\n", config->warning_sound);
|
|
ast_verbose(VERBOSE_PREFIX_4 "end_sound = %s\n", S_OR(config->end_sound, ""));
|
|
}
|
|
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);
|
|
return 0;
|
|
}
|
|
|
|
static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
|
|
struct ast_flags *opts, char **opt_args, struct privacy_args *pa)
|
|
{
|
|
|
|
int res2;
|
|
int loopcount = 0;
|
|
|
|
/* Get the user's intro, store it in priv-callerintros/$CID,
|
|
unless it is already there-- this should be done before the
|
|
call is actually dialed */
|
|
|
|
/* all ring indications and moh for the caller has been halted as soon as the
|
|
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 */
|
|
|
|
if (ast_test_flag(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
|
|
char *original_moh = ast_strdupa(chan->musicclass);
|
|
ast_indicate(chan, -1);
|
|
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 if (ast_test_flag(opts, OPT_RINGBACK)) {
|
|
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++) {
|
|
if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
|
|
break;
|
|
if (!res2) /* on timeout, play the message again */
|
|
res2 = ast_play_and_wait(peer,"priv-callpending");
|
|
if (!valid_priv_reply(opts, res2))
|
|
res2 = 0;
|
|
/* priv-callpending script:
|
|
"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 */
|
|
if( !res2 ) {
|
|
/* XXX can we have both, or they are mutually exclusive ? */
|
|
if( ast_test_flag(opts, OPT_PRIVACY) )
|
|
res2 = ast_play_and_wait(peer,"priv-callee-options");
|
|
if( ast_test_flag(opts, OPT_SCREENING) )
|
|
res2 = ast_play_and_wait(peer,"screen-callee-options");
|
|
}
|
|
/*! \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
|
|
Dial 2 if you wish to send this caller to voicemail now and
|
|
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");
|
|
}
|
|
|
|
if (ast_test_flag(opts, OPT_MUSICBACK)) {
|
|
ast_moh_stop(chan);
|
|
} else if (ast_test_flag(opts, OPT_RINGBACK)) {
|
|
ast_indicate(chan, -1);
|
|
pa->sentringing=0;
|
|
}
|
|
ast_autoservice_stop(chan);
|
|
if(ast_test_flag(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
|
|
/* map keypresses to various things, the index is res2 - '1' */
|
|
static const char *_val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
|
|
static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
|
|
int i = res2 - '1';
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to %s\n",
|
|
opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
|
|
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 ? */
|
|
if( ast_test_flag(opts, OPT_PRIVACY) )
|
|
break;
|
|
/* if not privacy, then 5 is the same as "default" case */
|
|
default: /* bad input or -1 if failure to start autoservice */
|
|
/* 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;
|
|
}
|
|
|
|
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
|
|
just clog things up, and it's not useful information, not being tied to a CID */
|
|
if( strncmp(pa->privcid,"NOCALLERID",10) == 0 || ast_test_flag(opts, OPT_SCREEN_NOINTRO) ) {
|
|
ast_filedelete(pa->privintro, NULL);
|
|
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);
|
|
else if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Successfully deleted %s intro file\n", pa->privintro);
|
|
}
|
|
return 0; /* the good exit path */
|
|
} else {
|
|
ast_hangup(peer); /* hang up on the callee -- he didn't want to talk anyway! */
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
/*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
|
|
static int setup_privacy_args(struct privacy_args *pa,
|
|
struct ast_flags *opts, char *opt_args[], struct ast_channel *chan)
|
|
{
|
|
char callerid[60];
|
|
int res;
|
|
char *l;
|
|
|
|
if (!ast_strlen_zero(chan->cid.cid_num)) {
|
|
l = ast_strdupa(chan->cid.cid_num);
|
|
ast_shrink_phone_number(l);
|
|
if (ast_test_flag(opts, OPT_PRIVACY) ) {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Privacy DB is '%s', clid is '%s'\n",
|
|
opt_args[OPT_ARG_PRIVACY], l);
|
|
pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
|
|
} else {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Privacy Screening, clid is '%s'\n", l);
|
|
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++) {
|
|
if (*tn2=='/') /* any other chars to be afraid of? */
|
|
*tn2 = '=';
|
|
}
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Privacy-- callerid is empty\n");
|
|
|
|
snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", chan->exten, tnam);
|
|
l = callerid;
|
|
pa->privdb_val = AST_PRIVACY_UNKNOWN;
|
|
}
|
|
|
|
ast_copy_string(pa->privcid,l,sizeof(pa->privcid));
|
|
|
|
if( strncmp(pa->privcid,"NOCALLERID",10) != 0 && ast_test_flag(opts, OPT_SCREEN_NOCLID) ) { /* if callerid is set, and ast_test_flag(&opts, OPT_SCREEN_NOCLID) is set also */
|
|
if (option_verbose > 2)
|
|
ast_verbose( VERBOSE_PREFIX_3 "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
|
|
pa->privdb_val = AST_PRIVACY_ALLOW;
|
|
} else if (ast_test_flag(opts, OPT_SCREEN_NOCLID) && strncmp(pa->privcid,"NOCALLERID",10) == 0 ) {
|
|
if (option_verbose > 2)
|
|
ast_verbose( VERBOSE_PREFIX_3 "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
|
|
}
|
|
|
|
if (pa->privdb_val == AST_PRIVACY_DENY ) {
|
|
ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
|
|
if (option_verbose > 2)
|
|
ast_verbose( VERBOSE_PREFIX_3 "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
|
|
return 0;
|
|
} else if (pa->privdb_val == AST_PRIVACY_KILL ) {
|
|
ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
|
|
if (ast_opt_priority_jumping || ast_test_flag(opts, OPT_PRIORITY_JUMP)) {
|
|
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 201);
|
|
}
|
|
return 0; /* Is this right? */
|
|
} else if (pa->privdb_val == AST_PRIVACY_TORTURE ) {
|
|
ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
|
|
if (ast_opt_priority_jumping || ast_test_flag(opts, OPT_PRIORITY_JUMP)) {
|
|
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 301);
|
|
}
|
|
return 0; /* is this right??? */
|
|
} else if (pa->privdb_val == AST_PRIVACY_UNKNOWN ) {
|
|
/* Get the user's intro, store it in priv-callerintros/$CID,
|
|
unless it is already there-- this should be done before the
|
|
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);
|
|
if (mkdir(pa->privintro, 0755) && errno != EEXIST) {
|
|
ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(errno));
|
|
return -1;
|
|
}
|
|
|
|
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) {
|
|
/* 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:"
|
|
|
|
*/
|
|
ast_answer(chan);
|
|
res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "gsm", &duration, 128, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
|
|
/* 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);
|
|
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);
|
|
else if (option_verbose > 2)
|
|
ast_verbose( VERBOSE_PREFIX_3 "Successfully deleted %s intro file\n", pa->privintro);
|
|
return -1;
|
|
}
|
|
if (!ast_streamfile(chan, "vm-dialout", chan->language) )
|
|
ast_waitstream(chan, "");
|
|
}
|
|
}
|
|
return 1; /* success */
|
|
}
|
|
|
|
static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags *peerflags, int *continue_exec)
|
|
{
|
|
int res = -1; /* default: error */
|
|
struct ast_module_user *u;
|
|
char *rest, *cur; /* scan the list of destinations */
|
|
struct chanlist *outgoing = NULL; /* list of destinations */
|
|
struct ast_channel *peer;
|
|
int to; /* timeout */
|
|
struct cause_args num = { chan, 0, 0, 0 };
|
|
int cause;
|
|
char numsubst[256];
|
|
char cidname[AST_MAX_EXTENSION] = "";
|
|
|
|
struct ast_bridge_config config;
|
|
unsigned int calldurationlimit = 0;
|
|
char *dtmfcalled = NULL, *dtmfcalling = NULL;
|
|
struct privacy_args pa = {
|
|
.sentringing = 0,
|
|
.privdb_val = 0,
|
|
.status = "INVALIDARGS",
|
|
};
|
|
int sentringing = 0, moh = 0;
|
|
const char *outbound_group = NULL;
|
|
int result = 0;
|
|
time_t start_time;
|
|
char *parse;
|
|
int opermode = 0;
|
|
AST_DECLARE_APP_ARGS(args,
|
|
AST_APP_ARG(peers);
|
|
AST_APP_ARG(timeout);
|
|
AST_APP_ARG(options);
|
|
AST_APP_ARG(url);
|
|
);
|
|
struct ast_flags opts = { 0, };
|
|
char *opt_args[OPT_ARG_ARRAY_SIZE];
|
|
|
|
if (ast_strlen_zero(data)) {
|
|
ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
|
|
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
|
|
return -1;
|
|
}
|
|
|
|
u = ast_module_user_add(chan); /* XXX is this the right place ? */
|
|
|
|
parse = ast_strdupa(data);
|
|
|
|
AST_STANDARD_APP_ARGS(args, parse);
|
|
|
|
memset(&config,0,sizeof(struct ast_bridge_config));
|
|
|
|
if (!ast_strlen_zero(args.options) &&
|
|
ast_app_parse_options(dial_exec_options, &opts, opt_args, args.options)) {
|
|
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
|
|
goto done;
|
|
}
|
|
|
|
if (ast_strlen_zero(args.peers)) {
|
|
ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
|
|
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
|
|
goto done;
|
|
}
|
|
|
|
if (ast_test_flag(&opts, OPT_OPERMODE)) {
|
|
opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Setting operator services mode to %d.\n", opermode);
|
|
}
|
|
|
|
if (ast_test_flag(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
|
|
calldurationlimit = atoi(opt_args[OPT_ARG_DURATION_STOP]);
|
|
if (!calldurationlimit) {
|
|
ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
|
|
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
|
|
goto done;
|
|
}
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Setting call duration limit to %d seconds.\n", calldurationlimit);
|
|
}
|
|
|
|
if (ast_test_flag(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
|
|
dtmfcalling = opt_args[OPT_ARG_SENDDTMF];
|
|
dtmfcalled = strsep(&dtmfcalling, ":");
|
|
}
|
|
|
|
if (ast_test_flag(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
|
|
if (do_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
|
|
goto done;
|
|
}
|
|
|
|
if (ast_test_flag(&opts, OPT_RESETCDR) && chan->cdr)
|
|
ast_cdr_reset(chan->cdr, NULL);
|
|
if (ast_test_flag(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
|
|
opt_args[OPT_ARG_PRIVACY] = ast_strdupa(chan->exten);
|
|
|
|
if (ast_test_flag(&opts, OPT_PRIVACY) || ast_test_flag(&opts, OPT_SCREENING)) {
|
|
res = setup_privacy_args(&pa, &opts, opt_args, chan);
|
|
if (res <= 0)
|
|
goto out;
|
|
res = -1; /* reset default */
|
|
}
|
|
|
|
if (continue_exec)
|
|
*continue_exec = 0;
|
|
|
|
/* If a channel group has been specified, get it for use when we create peer channels */
|
|
if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
|
|
outbound_group = ast_strdupa(outbound_group);
|
|
pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
|
|
} else {
|
|
outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP");
|
|
}
|
|
|
|
ast_copy_flags(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING);
|
|
/* loop through the list of dial destinations */
|
|
rest = args.peers;
|
|
while ((cur = strsep(&rest, "&")) ) {
|
|
struct chanlist *tmp;
|
|
struct ast_channel *tc; /* channel for this destination */
|
|
/* Get a technology/[device:]number pair */
|
|
char *number = cur;
|
|
char *tech = strsep(&number, "/");
|
|
if (!number) {
|
|
ast_log(LOG_WARNING, "Dial argument takes format (technology/[device:]number1)\n");
|
|
goto out;
|
|
}
|
|
if (!(tmp = ast_calloc(1, sizeof(*tmp))))
|
|
goto out;
|
|
if (opts.flags) {
|
|
ast_copy_flags(tmp, &opts,
|
|
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_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID);
|
|
ast_set2_flag(tmp, args.url, DIAL_NOFORWARDHTML);
|
|
}
|
|
ast_copy_string(numsubst, number, sizeof(numsubst));
|
|
/* Request the peer */
|
|
tc = ast_request(tech, chan->nativeformats, numsubst, &cause);
|
|
if (!tc) {
|
|
/* If we can't, just go on to the next call */
|
|
ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
|
|
tech, cause, ast_cause2str(cause));
|
|
handle_cause(cause, &num);
|
|
if (!rest) /* we are on the last destination */
|
|
chan->hangupcause = cause;
|
|
free(tmp);
|
|
continue;
|
|
}
|
|
pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", numsubst);
|
|
if (!ast_strlen_zero(tc->call_forward)) {
|
|
char tmpchan[256];
|
|
char *stuff;
|
|
char *tech;
|
|
ast_copy_string(tmpchan, tc->call_forward, sizeof(tmpchan));
|
|
if ((stuff = strchr(tmpchan, '/'))) {
|
|
*stuff++ = '\0';
|
|
tech = tmpchan;
|
|
} else {
|
|
snprintf(tmpchan, sizeof(tmpchan), "%s@%s", tc->call_forward, tc->context);
|
|
stuff = tmpchan;
|
|
tech = "Local";
|
|
}
|
|
tmp->forwards++;
|
|
if (tmp->forwards < AST_MAX_FORWARDS) {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Now forwarding %s to '%s/%s' (thanks to %s)\n",
|
|
chan->name, tech, stuff, tc->name);
|
|
ast_hangup(tc);
|
|
/* If we have been told to ignore forwards, just set this channel to null
|
|
* and continue processing extensions normally */
|
|
if (ast_test_flag(&opts, OPT_IGNORE_FORWARDING)) {
|
|
tc = NULL;
|
|
cause = AST_CAUSE_BUSY;
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Forwarding %s to '%s/%s' prevented.\n",
|
|
chan->name, tech, stuff);
|
|
} else {
|
|
tc = ast_request(tech, chan->nativeformats, stuff, &cause);
|
|
}
|
|
if (!tc)
|
|
ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, cause);
|
|
else
|
|
ast_channel_inherit_variables(chan, tc);
|
|
} else {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Too many forwards from %s\n", tc->name);
|
|
ast_hangup(tc);
|
|
tc = NULL;
|
|
cause = AST_CAUSE_CONGESTION;
|
|
}
|
|
if (!tc) {
|
|
handle_cause(cause, &num);
|
|
free(tmp);
|
|
continue;
|
|
}
|
|
}
|
|
|
|
/* Setup outgoing SDP to match incoming one */
|
|
ast_rtp_make_compatible(tc, chan, !outgoing && !rest);
|
|
|
|
/* Inherit specially named variables from parent channel */
|
|
ast_channel_inherit_variables(chan, tc);
|
|
|
|
tc->appl = "AppDial";
|
|
tc->data = "(Outgoing Line)";
|
|
tc->whentohangup = 0;
|
|
|
|
S_REPLACE(tc->cid.cid_num, ast_strdup(chan->cid.cid_num));
|
|
S_REPLACE(tc->cid.cid_name, ast_strdup(chan->cid.cid_name));
|
|
S_REPLACE(tc->cid.cid_ani, ast_strdup(chan->cid.cid_ani));
|
|
S_REPLACE(tc->cid.cid_rdnis, ast_strdup(chan->cid.cid_rdnis));
|
|
|
|
/* Copy language from incoming to outgoing */
|
|
ast_string_field_set(tc, language, chan->language);
|
|
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);
|
|
/* Pass callingpres, type of number, tns, ADSI CPE, transfer capability */
|
|
tc->cid.cid_pres = chan->cid.cid_pres;
|
|
tc->cid.cid_ton = chan->cid.cid_ton;
|
|
tc->cid.cid_tns = chan->cid.cid_tns;
|
|
tc->adsicpe = chan->adsicpe;
|
|
tc->transfercapability = chan->transfercapability;
|
|
|
|
/* If we have an outbound group, set this peer channel to it */
|
|
if (outbound_group)
|
|
ast_app_group_set_channel(tc, outbound_group);
|
|
|
|
/* Inherit context and extension */
|
|
if (!ast_strlen_zero(chan->macrocontext))
|
|
ast_copy_string(tc->dialcontext, chan->macrocontext, sizeof(tc->dialcontext));
|
|
else
|
|
ast_copy_string(tc->dialcontext, chan->context, sizeof(tc->dialcontext));
|
|
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));
|
|
|
|
res = ast_call(tc, numsubst, 0); /* Place the call, but don't wait on the answer */
|
|
|
|
/* Save the info in cdr's that we called them */
|
|
if (chan->cdr)
|
|
ast_cdr_setdestchan(chan->cdr, tc->name);
|
|
|
|
/* check the results of ast_call */
|
|
if (res) {
|
|
/* Again, keep going even if there's an error */
|
|
if (option_debug)
|
|
ast_log(LOG_DEBUG, "ast call on peer returned %d\n", res);
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Couldn't call %s\n", numsubst);
|
|
ast_hangup(tc);
|
|
tc = NULL;
|
|
free(tmp);
|
|
continue;
|
|
} else {
|
|
senddialevent(chan, tc);
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Called %s\n", numsubst);
|
|
if (!ast_test_flag(peerflags, OPT_ORIGINAL_CLID))
|
|
ast_set_callerid(tc, S_OR(chan->macroexten, chan->exten), get_cid_name(cidname, sizeof(cidname), chan), NULL);
|
|
}
|
|
/* Put them in the list of outgoing thingies... We're ready now.
|
|
XXX If we're forcibly removed, these outgoing calls won't get
|
|
hung up XXX */
|
|
ast_set_flag(tmp, DIAL_STILLGOING);
|
|
tmp->chan = tc;
|
|
tmp->next = outgoing;
|
|
outgoing = tmp;
|
|
/* If this line is up, don't try anybody else */
|
|
if (outgoing->chan->_state == AST_STATE_UP)
|
|
break;
|
|
}
|
|
|
|
if (ast_strlen_zero(args.timeout)) {
|
|
to = -1;
|
|
} else {
|
|
to = atoi(args.timeout);
|
|
if (to > 0)
|
|
to *= 1000;
|
|
else
|
|
ast_log(LOG_WARNING, "Invalid timeout specified: '%s'\n", args.timeout);
|
|
}
|
|
|
|
if (!outgoing) {
|
|
strcpy(pa.status, "CHANUNAVAIL");
|
|
} else {
|
|
/* Our status will at least be NOANSWER */
|
|
strcpy(pa.status, "NOANSWER");
|
|
if (ast_test_flag(outgoing, OPT_MUSICBACK)) {
|
|
moh = 1;
|
|
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);
|
|
}
|
|
ast_indicate(chan, AST_CONTROL_PROGRESS);
|
|
} else if (ast_test_flag(outgoing, OPT_RINGBACK)) {
|
|
ast_indicate(chan, AST_CONTROL_RINGING);
|
|
sentringing++;
|
|
}
|
|
}
|
|
|
|
time(&start_time);
|
|
peer = wait_for_answer(chan, outgoing, &to, peerflags, &pa, &num, ast_test_flag(&opts, OPT_PRIORITY_JUMP), &result);
|
|
|
|
if (!peer) {
|
|
if (result) {
|
|
res = result;
|
|
} else if (to) { /* Musta gotten hung up */
|
|
res = -1;
|
|
} else { /* Nobody answered, next please? */
|
|
res = 0;
|
|
}
|
|
/* almost done, although the 'else' block is 400 lines */
|
|
} else {
|
|
const char *number;
|
|
time_t end_time, answer_time = time(NULL);
|
|
char toast[80]; /* buffer to set variables */
|
|
|
|
strcpy(pa.status, "ANSWER");
|
|
/* Ah ha! Someone answered within the desired timeframe. Of course after this
|
|
we will always return with -1 so that it is hung up properly after the
|
|
conversation. */
|
|
hanguptree(outgoing, peer);
|
|
outgoing = NULL;
|
|
/* If appropriate, log that we have a destination channel */
|
|
if (chan->cdr)
|
|
ast_cdr_setdestchan(chan->cdr, peer->name);
|
|
if (peer->name)
|
|
pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
|
|
|
|
number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
|
|
if (!number)
|
|
number = numsubst;
|
|
pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
|
|
if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
|
|
if (option_debug)
|
|
ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", args.url);
|
|
ast_channel_sendurl( peer, args.url );
|
|
}
|
|
if ( (ast_test_flag(&opts, OPT_PRIVACY) || ast_test_flag(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
|
|
if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
|
|
res = 0;
|
|
goto out;
|
|
}
|
|
}
|
|
if (!ast_test_flag(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
|
|
res = 0;
|
|
} else {
|
|
int digit = 0;
|
|
/* Start autoservice on the other chan */
|
|
res = ast_autoservice_start(chan);
|
|
/* Now Stream the File */
|
|
if (!res)
|
|
res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], peer->language);
|
|
if (!res) {
|
|
digit = ast_waitstream(peer, AST_DIGIT_ANY);
|
|
}
|
|
/* Ok, done. stop autoservice */
|
|
res = ast_autoservice_stop(chan);
|
|
if (digit > 0 && !res)
|
|
res = ast_senddigit(chan, digit);
|
|
else
|
|
res = digit;
|
|
|
|
}
|
|
|
|
if (chan && peer && ast_test_flag(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
|
|
replace_macro_delimiter(opt_args[OPT_ARG_GOTO]);
|
|
ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
|
|
ast_parseable_goto(peer, opt_args[OPT_ARG_GOTO]);
|
|
peer->priority++;
|
|
ast_pbx_start(peer);
|
|
hanguptree(outgoing, NULL);
|
|
if (continue_exec)
|
|
*continue_exec = 1;
|
|
res = 0;
|
|
goto done;
|
|
}
|
|
|
|
if (ast_test_flag(&opts, OPT_CALLEE_MACRO) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_MACRO])) {
|
|
struct ast_app *theapp;
|
|
const char *macro_result;
|
|
|
|
res = ast_autoservice_start(chan);
|
|
if (res) {
|
|
ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
|
|
res = -1;
|
|
}
|
|
|
|
theapp = pbx_findapp("Macro");
|
|
|
|
if (theapp && !res) { /* XXX why check res here ? */
|
|
replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
|
|
res = pbx_exec(peer, theapp, opt_args[OPT_ARG_CALLEE_MACRO]);
|
|
if (option_debug)
|
|
ast_log(LOG_DEBUG, "Macro exited with status %d\n", res);
|
|
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;
|
|
}
|
|
|
|
if (!res && (macro_result = pbx_builtin_getvar_helper(peer, "MACRO_RESULT"))) {
|
|
char *macro_transfer_dest;
|
|
|
|
if (!strcasecmp(macro_result, "BUSY")) {
|
|
ast_copy_string(pa.status, macro_result, sizeof(pa.status));
|
|
if (ast_opt_priority_jumping || ast_test_flag(&opts, OPT_PRIORITY_JUMP)) {
|
|
if (!ast_goto_if_exists(chan, NULL, NULL, chan->priority + 101)) {
|
|
ast_set_flag(peerflags, OPT_GO_ON);
|
|
}
|
|
} else
|
|
ast_set_flag(peerflags, OPT_GO_ON);
|
|
res = -1;
|
|
} else if (!strcasecmp(macro_result, "CONGESTION") || !strcasecmp(macro_result, "CHANUNAVAIL")) {
|
|
ast_copy_string(pa.status, macro_result, sizeof(pa.status));
|
|
ast_set_flag(peerflags, OPT_GO_ON);
|
|
res = -1;
|
|
} else if (!strcasecmp(macro_result, "CONTINUE")) {
|
|
/* hangup peer and keep chan alive assuming the macro has changed
|
|
the context / exten / priority or perhaps
|
|
the next priority in the current exten is desired.
|
|
*/
|
|
ast_set_flag(peerflags, OPT_GO_ON);
|
|
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))
|
|
ast_set_flag(peerflags, OPT_GO_ON);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (!res) {
|
|
if (calldurationlimit > 0) {
|
|
chan->whentohangup = time(NULL) + calldurationlimit;
|
|
}
|
|
if (!ast_strlen_zero(dtmfcalled)) {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Sending DTMF '%s' to the called party.\n", dtmfcalled);
|
|
res = ast_dtmf_stream(peer,chan,dtmfcalled,250);
|
|
}
|
|
if (!ast_strlen_zero(dtmfcalling)) {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
|
|
res = ast_dtmf_stream(chan,peer,dtmfcalling,250);
|
|
}
|
|
}
|
|
|
|
if (res) { /* some error */
|
|
res = -1;
|
|
end_time = time(NULL);
|
|
} else {
|
|
if (ast_test_flag(peerflags, OPT_CALLEE_TRANSFER))
|
|
ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
|
|
if (ast_test_flag(peerflags, OPT_CALLER_TRANSFER))
|
|
ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
|
|
if (ast_test_flag(peerflags, OPT_CALLEE_HANGUP))
|
|
ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
|
|
if (ast_test_flag(peerflags, OPT_CALLER_HANGUP))
|
|
ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
|
|
if (ast_test_flag(peerflags, OPT_CALLEE_MONITOR))
|
|
ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
|
|
if (ast_test_flag(peerflags, OPT_CALLER_MONITOR))
|
|
ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
|
|
if (ast_test_flag(peerflags, OPT_CALLEE_PARK))
|
|
ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
|
|
if (ast_test_flag(peerflags, OPT_CALLER_PARK))
|
|
ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
|
|
|
|
if (moh) {
|
|
moh = 0;
|
|
ast_moh_stop(chan);
|
|
} else if (sentringing) {
|
|
sentringing = 0;
|
|
ast_indicate(chan, -1);
|
|
}
|
|
/* 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);
|
|
res = -1;
|
|
goto done;
|
|
}
|
|
if (opermode && !strncmp(chan->name,"Zap",3) && !strncmp(peer->name,"Zap",3)) {
|
|
/* XXX what's this special handling for Zap <-> Zap ? */
|
|
struct oprmode oprmode;
|
|
|
|
oprmode.peer = peer;
|
|
oprmode.mode = opermode;
|
|
|
|
ast_channel_setoption(chan,
|
|
AST_OPTION_OPRMODE,&oprmode,sizeof(struct oprmode),0);
|
|
}
|
|
res = ast_bridge_call(chan,peer,&config);
|
|
end_time = time(NULL);
|
|
snprintf(toast, sizeof(toast), "%ld", (long)(end_time - answer_time));
|
|
pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", toast);
|
|
}
|
|
|
|
snprintf(toast, sizeof(toast), "%ld", (long)(end_time - start_time));
|
|
pbx_builtin_setvar_helper(chan, "DIALEDTIME", toast);
|
|
|
|
if (res != AST_PBX_NO_HANGUP_PEER) {
|
|
if (!chan->_softhangup)
|
|
chan->hangupcause = peer->hangupcause;
|
|
ast_hangup(peer);
|
|
}
|
|
}
|
|
out:
|
|
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);
|
|
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
|
|
senddialendevent(chan, pa.status);
|
|
if (option_debug)
|
|
ast_log(LOG_DEBUG, "Exiting with DIALSTATUS=%s.\n", pa.status);
|
|
|
|
if ((ast_test_flag(peerflags, OPT_GO_ON)) && (!chan->_softhangup) && (res != AST_PBX_KEEPALIVE))
|
|
res = 0;
|
|
|
|
done:
|
|
ast_module_user_remove(u); /* XXX probably not the right place for this. */
|
|
return res;
|
|
}
|
|
|
|
static int dial_exec(struct ast_channel *chan, void *data)
|
|
{
|
|
struct ast_flags peerflags;
|
|
|
|
memset(&peerflags, 0, sizeof(peerflags));
|
|
|
|
return dial_exec_full(chan, data, &peerflags, NULL);
|
|
}
|
|
|
|
static int retrydial_exec(struct ast_channel *chan, void *data)
|
|
{
|
|
char *announce = NULL, *dialdata = NULL;
|
|
const char *context = NULL;
|
|
int sleep = 0, loops = 0, res = -1;
|
|
struct ast_module_user *u;
|
|
struct ast_flags peerflags;
|
|
|
|
if (ast_strlen_zero(data)) {
|
|
ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
|
|
return -1;
|
|
}
|
|
|
|
u = ast_module_user_add(chan);
|
|
|
|
announce = ast_strdupa(data);
|
|
|
|
memset(&peerflags, 0, sizeof(peerflags));
|
|
|
|
if ((dialdata = strchr(announce, '|'))) {
|
|
*dialdata++ = '\0';
|
|
if ((sleep = atoi(dialdata))) {
|
|
sleep *= 1000;
|
|
} else {
|
|
ast_log(LOG_ERROR, "%s requires the numerical argument <sleep>\n",rapp);
|
|
goto done;
|
|
}
|
|
if ((dialdata = strchr(dialdata, '|'))) {
|
|
*dialdata++ = '\0';
|
|
if (!(loops = atoi(dialdata))) {
|
|
ast_log(LOG_ERROR, "%s requires the numerical argument <loops>\n",rapp);
|
|
goto done;
|
|
}
|
|
}
|
|
}
|
|
|
|
if ((dialdata = strchr(dialdata, '|'))) {
|
|
*dialdata++ = '\0';
|
|
} else {
|
|
ast_log(LOG_ERROR, "%s requires more arguments\n",rapp);
|
|
goto done;
|
|
}
|
|
|
|
if (sleep < 1000)
|
|
sleep = 10000;
|
|
|
|
if (!loops)
|
|
loops = -1; /* run forever */
|
|
|
|
context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
|
|
|
|
res = 0;
|
|
while (loops) {
|
|
int continue_exec;
|
|
|
|
chan->data = "Retrying";
|
|
if (ast_test_flag(chan, AST_FLAG_MOH))
|
|
ast_moh_stop(chan);
|
|
|
|
res = dial_exec_full(chan, dialdata, &peerflags, &continue_exec);
|
|
if (continue_exec)
|
|
break;
|
|
if (res == 0) {
|
|
if (ast_test_flag(&peerflags, OPT_DTMF_EXIT)) {
|
|
if (!(res = ast_streamfile(chan, announce, chan->language)))
|
|
res = ast_waitstream(chan, AST_DIGIT_ANY);
|
|
if (!res && sleep) {
|
|
if (!ast_test_flag(chan, AST_FLAG_MOH))
|
|
ast_moh_start(chan, NULL, NULL);
|
|
res = ast_waitfordigit(chan, sleep);
|
|
}
|
|
} else {
|
|
if (!(res = ast_streamfile(chan, announce, chan->language)))
|
|
res = ast_waitstream(chan, "");
|
|
if (sleep) {
|
|
if (!ast_test_flag(chan, AST_FLAG_MOH))
|
|
ast_moh_start(chan, NULL, NULL);
|
|
if (!res)
|
|
res = ast_waitfordigit(chan, sleep);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (res < 0)
|
|
break;
|
|
else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
|
|
if (onedigit_goto(chan, context, (char) res, 1)) {
|
|
res = 0;
|
|
break;
|
|
}
|
|
}
|
|
loops--;
|
|
}
|
|
if (loops == 0)
|
|
res = 0;
|
|
else if (res == 1)
|
|
res = 0;
|
|
|
|
if (ast_test_flag(chan, AST_FLAG_MOH))
|
|
ast_moh_stop(chan);
|
|
done:
|
|
ast_module_user_remove(u);
|
|
return res;
|
|
}
|
|
|
|
static int unload_module(void)
|
|
{
|
|
int res;
|
|
|
|
res = ast_unregister_application(app);
|
|
res |= ast_unregister_application(rapp);
|
|
|
|
ast_module_user_hangup_all();
|
|
|
|
return res;
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
int res;
|
|
|
|
res = ast_register_application(app, dial_exec, synopsis, descrip);
|
|
res |= ast_register_application(rapp, retrydial_exec, rsynopsis, rdescrip);
|
|
|
|
return res;
|
|
}
|
|
|
|
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Dialing Application");
|